Re: [asterisk-users] Asterisk Realtime Billing Question???
Hi Sherwood ,
well , i think you did not understand my question , i want real time billing
like as i mentioned that if i want to dial 5 number with different call rate
how can i access same
balance into those 5 people, if all are connected how can i periodically
update billing , as you suggested it will assign total balance to those 5
people but actually we can not do like this as total balance of user $100 ,
as per your suggestion it will give $100 for those 5 people which is
practically wrong i think.
give your thougts.
regards
dhaval
On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hello All,
after so long time i posted a new question regarding billing, hope anyone
have some solution.
I have situation in that i want to do billing of more than 1 call in real
time below are scenario and explanation.
Scenario:
A customer called my DID number and after that from here i dial few
number let say 5 number. once number are placed into DIAL
i will put this customer into conference [MEETME] , once a Members are
picked up call they will also patched into conference and
talking is started, every thing working fine with DIAL-PLAN and DB look
up.
Now, i want to do billing on customer dialed my DID, and from that
actually it DIALED 5 numbers, how can i DO real time billing
into this situation, like numbers can be different It can be ISD,STD,Local
and also free .
if customer having initial balance of $100 then how can i check balance
every time.in a situation once balance is nil then i want to disconnect
calls . is any one facing this type of situation.
give me some idea ,
regards
Dhaval
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Dhaval,
This sounds very much like a system I'm working on for a client right now.
I'm not permitted to disclose much about it due to the NDA i signed, but
I'll risk giving you a point in the right direction.
First, you should create a table in your database that has a column called
callid, and other columns that you will have to decide upon. This table will
be called something like '*call_references*'. Oh, and you'll want to
define callid as the primary key for records in that table, but DO NOT make
it an autoincrement, you're going to populate it with a value that is
described in the next step.
Second, at the beginning of the original call you mentioned, define a
variable that will be unique to that call. I personally have done this by
stripping all non-digits from the caller's callerid (using
Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.
Next (this is where I have to start being a bit vague), you're going to
perform an INSERT query, creating a new call_references record (using that
variable I just showed you how to construct as callid's value).
Now, when you defined that variable, you should have preceded the variable
name with two underscores ( __ ), which will tell Asterisk that channels
spawned by the current channel will inherit that variable and it's value.
Voila, you now have a method for storing realtime data such as billing
information between MULTIPLE calls.
I wish I could tell you more, but I can't violate my client's
Non-Disclosure Agreement.
Hope this helps you out!
Sherwood McGowan
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Re: [asterisk-users] Recommendation for a new server
On Thu, 2010-10-21 at 01:15 -0400, Zeeshan Zakaria wrote: Yes, one server will do it all. It will not be in a data center but at customer premisis, so doesn't have to be 1U. In that case, how about a dell-server? And if it is not in a data center, take care of an UPS for both the server and any network equipment (modem's, switches, POE-phones) hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime Billing Question???
On Thu, Oct 21, 2010 at 1:30 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:
Hi Sherwood ,
well , i think you did not understand my question , i want real time
billing
like as i mentioned that if i want to dial 5 number with different call
rate how can i access same
balance into those 5 people, if all are connected how can i periodically
update billing , as you suggested it will assign total balance to those 5
people but actually we can not do like this as total balance of user $100 ,
as per your suggestion it will give $100 for those 5 people which is
practically wrong i think.
give your thougts.
regards
dhaval
On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hello All,
after so long time i posted a new question regarding billing, hope
anyone have some solution.
I have situation in that i want to do billing of more than 1 call in real
time below are scenario and explanation.
Scenario:
A customer called my DID number and after that from here i dial few
number let say 5 number. once number are placed into DIAL
i will put this customer into conference [MEETME] , once a Members are
picked up call they will also patched into conference and
talking is started, every thing working fine with DIAL-PLAN and DB look
up.
Now, i want to do billing on customer dialed my DID, and from that
actually it DIALED 5 numbers, how can i DO real time billing
into this situation, like numbers can be different It can be
ISD,STD,Local and also free .
if customer having initial balance of $100 then how can i check balance
every time.in a situation once balance is nil then i want to disconnect
calls . is any one facing this type of situation.
give me some idea ,
regards
Dhaval
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asterisk-users mailing list
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Dhaval,
This sounds very much like a system I'm working on for a client right now.
I'm not permitted to disclose much about it due to the NDA i signed, but
I'll risk giving you a point in the right direction.
First, you should create a table in your database that has a column called
callid, and other columns that you will have to decide upon. This table will
be called something like '*call_references*'. Oh, and you'll want to
define callid as the primary key for records in that table, but DO NOT make
it an autoincrement, you're going to populate it with a value that is
described in the next step.
Second, at the beginning of the original call you mentioned, define a
variable that will be unique to that call. I personally have done this by
stripping all non-digits from the caller's callerid (using
Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.
Next (this is where I have to start being a bit vague), you're going to
perform an INSERT query, creating a new call_references record (using that
variable I just showed you how to construct as callid's value).
Now, when you defined that variable, you should have preceded the variable
name with two underscores ( __ ), which will tell Asterisk that channels
spawned by the current channel will inherit that variable and it's value.
Voila, you now have a method for storing realtime data such as billing
information between MULTIPLE calls.
I wish I could tell you more, but I can't violate my client's
Non-Disclosure Agreement.
Hope this helps you out!
Sherwood McGowan
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asterisk-users mailing list
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Well, you got the right guy, I've written several different RT billing
setups for clients ranging from small residential ITSPs all the way up to a
wholesale carrier in Austria. . .
What you'd have to do is have a column called freeze in your table that
you keep customer accounts and billing info in
Re: [asterisk-users] Error with Connecting Two Asterisk BOX with IAX
Hi bakko, just as a test, try to add calltokenoptional=0.0.0.0/0.0.0.0 to your iax.conf. Giorgio Incantalupo bakko wrote: Hello, I'm trying to conect two 1.6.2.13 Asterisk server with IAX. This is my configuration: Asterisk A: iax.conf register = coiax:pa...@69.164.207.166 [smiax] type=friend host=dynamic trunk=yes secret=pass2 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.207.166/255.255.255.255 qualify=yes Console: iax2 registry 69.164.207.166:4569 N coiax 69.164.197.105:456960 Registered iax2 peers smiax69.164.207.166 (D) 255.255.255.255 4569 (T) OK (3 ms) Asterisk B: register = smiax:pa...@69.164.197.105 [coiax] type=friend host=dynamic trunk=yes secret=pass1 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.197.105/255.255.255.255 qualify=yes Console iax2 registry 69.164.197.105:4569 N smiax 69.164.207.166:456960 Registered iax2 peers coiax69.164.197.105 (D) 255.255.255.255 4569 (T) OK (3 ms) When I try to call from Asterisk A to Asterisk B I receive this error Asterisk A WARNING[28759]: chan_iax2.c:10287 socket_process: Call rejected by 69.164.207.166: No authority found AsteriskB NOTICE[26563]: chan_iax2.c:10522 socket_process: Host 69.164.197.105 failed to authenticate as coiax What's wrong? Thank you in advance. Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue member status - BUSY
Have you tried playing with joinempty and leavewhenemèpty to avoid people being connected to a queue with all agents in use? l. 2010/10/20 GBR Icasiano, Ryan A. raicasi...@globalbridgeresources.com Hi, Is there a way to know if a member of a queue is currently engaged on a call? Or if a queue can return a busy status if all members are currently engaged in a call? QUEUESTATUS only returns FULL and TIMEOUT, and the scenario only falls into TIMEOUT, and has to finish the assigned number of seconds into the QUEUE CMD before it falls back to the next routine on the dialplan. Any ideas? regards, ryan icasiano -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DIALSTATUS always returns NOANSWER
Hi, Here is the scenario: 1. 1st phone calls and asterisk dials to extension no. 2. Extension answers 1st caller(which makes it busy). 2. 2nd phone calls and asterisk dials to extension no. 3. 2nd phone hears a BUSY tone, but have to wait for the timeout to expire(in DIAL cmd) before proceeding to the next step in dialplan 4. Get the current DIALSTATUS, but it returns NOANSWER, instead of BUSY the problem is, since the 2nd caller hears a busy tone, it should not wait for the timeout to expire, and proceed immediately in fetching the DIALSTATUS. I also tried this scenario and used DEV_STATE, but it always returns NOT_INUSE I already assigned qualify=yes in my sip configuration but still to no avail. any ideas? regards, RYAN ICASIANO -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime Billing Question???
thanks mate,
for useful and good information provided by you, i am not asking you that
please write down your all LOGIC and explain everything to me, as per your
explanation i can see it will deduct amount for only 1 call but what
actually i am searching for is if user made 5 concurrent calls and i have to
limit
all calls and each destination number having different rate may be some of
them ISD and some of them local. that will create more problem to me, i
think there is some solutions for this . could you suggest any reference for
the same, it will be more helpful to me.
thanks in advance,
regards
Dhaval
On Thu, Oct 21, 2010 at 12:49 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On Thu, Oct 21, 2010 at 1:30 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hi Sherwood ,
well , i think you did not understand my question , i want real time
billing
like as i mentioned that if i want to dial 5 number with different call
rate how can i access same
balance into those 5 people, if all are connected how can i periodically
update billing , as you suggested it will assign total balance to those 5
people but actually we can not do like this as total balance of user $100 ,
as per your suggestion it will give $100 for those 5 people which is
practically wrong i think.
give your thougts.
regards
dhaval
On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hello All,
after so long time i posted a new question regarding billing, hope
anyone have some solution.
I have situation in that i want to do billing of more than 1 call in
real time below are scenario and explanation.
Scenario:
A customer called my DID number and after that from here i dial few
number let say 5 number. once number are placed into DIAL
i will put this customer into conference [MEETME] , once a Members are
picked up call they will also patched into conference and
talking is started, every thing working fine with DIAL-PLAN and DB look
up.
Now, i want to do billing on customer dialed my DID, and from that
actually it DIALED 5 numbers, how can i DO real time billing
into this situation, like numbers can be different It can be
ISD,STD,Local and also free .
if customer having initial balance of $100 then how can i check balance
every time.in a situation once balance is nil then i want to disconnect
calls . is any one facing this type of situation.
give me some idea ,
regards
Dhaval
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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Dhaval,
This sounds very much like a system I'm working on for a client right
now. I'm not permitted to disclose much about it due to the NDA i signed,
but I'll risk giving you a point in the right direction.
First, you should create a table in your database that has a column
called callid, and other columns that you will have to decide upon. This
table will be called something like '*call_references*'. Oh, and you'll
want to define callid as the primary key for records in that table, but DO
NOT make it an autoincrement, you're going to populate it with a value that
is described in the next step.
Second, at the beginning of the original call you mentioned, define a
variable that will be unique to that call. I personally have done this by
stripping all non-digits from the caller's callerid (using
Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.
Next (this is where I have to start being a bit vague), you're going to
perform an INSERT query, creating a new call_references record (using that
variable I just showed you how to construct as callid's value).
Now, when you defined that variable, you should have preceded the
variable name with two underscores ( __ ), which will tell Asterisk that
channels spawned by the current channel will inherit that variable and it's
value.
Voila, you now have a method for storing realtime data such as billing
information between MULTIPLE calls.
I wish I could tell you more, but I can't violate my client's
Non-Disclosure Agreement.
Hope this helps you out!
Sherwood McGowan
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asterisk-users mailing list
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Re: [asterisk-users] Queue member status - BUSY
Hi, I didn't use that feature since i only added the phones not treated as agents(it will just ring the members, depending on the scenario chosen, instead of ringing the queue itself until an agent answers). The queue status is correct, although it could not tell if all members in the queue are currently engaged in a call(busy). I appreciate your help. Thanks! regards, RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri [lenz.lo...@gmail.com] Sent: Thursday, October 21, 2010 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue member status - BUSY Have you tried playing with joinempty and leavewhenemèpty to avoid people being connected to a queue with all agents in use? l. 2010/10/20 GBR Icasiano, Ryan A. raicasi...@globalbridgeresources.commailto:raicasi...@globalbridgeresources.com Hi, Is there a way to know if a member of a queue is currently engaged on a call? Or if a queue can return a busy status if all members are currently engaged in a call? QUEUESTATUS only returns FULL and TIMEOUT, and the scenario only falls into TIMEOUT, and has to finish the assigned number of seconds into the QUEUE CMD before it falls back to the next routine on the dialplan. Any ideas? regards, ryan icasiano -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial Plan Conf
Here I am expecting to be configured following scenario: User calls : it will play a sound will ask for input DTMF, then call will be given to particular extension for any DTMF entered. But its not working as expected. I have attached the dial plan file. 1.vdp Description: Binary data -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime Billing Question???
On Thu, Oct 21, 2010 at 3:23 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:
thanks mate,
for useful and good information provided by you, i am not asking you that
please write down your all LOGIC and explain everything to me, as per your
explanation i can see it will deduct amount for only 1 call but what
actually i am searching for is if user made 5 concurrent calls and i have to
limit
all calls and each destination number having different rate may be some of
them ISD and some of them local. that will create more problem to me, i
think there is some solutions for this . could you suggest any reference for
the same, it will be more helpful to me.
thanks in advance,
regards
Dhaval
*snip*
Dhaval,
This sounds very much like a system I'm working on for a client right
now. I'm not permitted to disclose much about it due to the NDA i signed,
but I'll risk giving you a point in the right direction.
First, you should create a table in your database that has a column
called callid, and other columns that you will have to decide upon. This
table will be called something like '*call_references*'. Oh, and you'll
want to define callid as the primary key for records in that table, but DO
NOT make it an autoincrement, you're going to populate it with a value that
is described in the next step.
Second, at the beginning of the original call you mentioned, define a
variable that will be unique to that call. I personally have done this by
stripping all non-digits from the caller's callerid (using
Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.
Next (this is where I have to start being a bit vague), you're going to
perform an INSERT query, creating a new call_references record (using that
variable I just showed you how to construct as callid's value).
Now, when you defined that variable, you should have preceded the
variable name with two underscores ( __ ), which will tell Asterisk that
channels spawned by the current channel will inherit that variable and it's
value.
Voila, you now have a method for storing realtime data such as billing
information between MULTIPLE calls.
I wish I could tell you more, but I can't violate my client's
Non-Disclosure Agreement.
Hope this helps you out!
Sherwood McGowan
Well, you got the right guy, I've written several different RT billing
setups for clients ranging from small residential ITSPs all the way up to a
wholesale carrier in Austria. . .
What you'd have to do is have a column called freeze in your table that
you keep customer accounts and billing info in (mainly, the balance). Then,
you'd need a 'frozen' table, with three columns: id, accountid (or some
other name/reference that references the customer in question), and amount.
Now, the freeze column in the account table defines how many minutes
worth of funds (at the rate the call is being charged to the customer)
you're going to make unavailable to the customer until the call is
completed. You multiply the value from freeze against the rate the call is
going to be charged at, resulting in amount_to_freeze. Subtract that
number from the customer's current balance, and then create a record in the
frozen table with that customer's accountid and put the value of
amount_to_freeze into the amount column.
Finally,when the customer's call(s) completes, calculate the total charge
for the call, check to see if it's more than `frozen`.`amount`, and if it
is, subtract `frozen`.`amount` from the total charge, and then subtract the
remaining amount from the customer's balance. If the total is *not* more
than than `frozen`.`amount`, you'll subtract total from `frozen`.`amount`,
and then ADDING the remaining amount to the customer's balance. (Being the
doofus I am, I called that procedure thawing, LOL)
In addition to the freezing of funds, you'll need to perform some magic
and limit the length of the customer's calls based on the balance of the
account just before freezing funds. This will need to be in conjunction with
having a maximum number of concurrent calls the customer can have, and
taking that into account when limiting each call.
It sounds complicated but I wrote this type of system several times, the
first couple were native to Asterisk using AELv2 (no AGI calls, more secure,
less resources hogged, etc), and then I wrote the last one using MySQL
stored procedures to perform just about ALL of the calculations and logic.
Basically at the beginning of a call, Asterisk would execute a stored
procedure called something like freeze_and_limit and passing two arguments,
the accountid and the rate per minute their call is going to cost (could
have also just fed the SP the destination and let it calculate THAT too).
MySQL would return the number of milliseconds the customer's call could be.
Alright, hopefully THAT gets you heading in the right
Re: [asterisk-users] Dial Plan Conf
On Thu, Oct 21, 2010 at 02:46:16PM +0530, Jigar Joshi wrote: Here I am expecting to be configured following scenario: User calls : it will play a sound will ask for input DTMF, then call will be given to particular extension for any DTMF entered. But its not working as expected. I have attached the dial plan file. Sorry, but I can't parse that VDP (Visual DialPlan?). -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Plan Conf
On 21 Oct 2010, at 10:16, Jigar Joshi wrote: I have attached the dial plan file. In what format? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime Billing Question???
If you look at it the way you want it.. you usually tell your customer the
available funds and minutes in their account right?
How will you explain politely that you have dropped their calls for lack
of balance because someone else used their account?
If you don't tell them their balance and call duration before call .. then
that won't be a problem.
Now you can do some kind of script to do the math and disconnect calls when
balance is over.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, October 21, 2010 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Realtime Billing Question???
Hi Sherwood ,
well , i think you did not understand my question , i want real time billing
like as i mentioned that if i want to dial 5 number with different call rate
how can i access same
balance into those 5 people, if all are connected how can i periodically
update billing , as you suggested it will assign total balance to those 5
people but actually we can not do like this as total balance of user $100 ,
as per your suggestion it will give $100 for those 5 people which is
practically wrong i think.
give your thougts.
regards
dhaval
On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hello All,
after so long time i posted a new question regarding billing, hope anyone
have some solution.
I have situation in that i want to do billing of more than 1 call in real
time below are scenario and explanation.
Scenario:
A customer called my DID number and after that from here i dial few number
let say 5 number. once number are placed into DIAL
i will put this customer into conference [MEETME] , once a Members are
picked up call they will also patched into conference and
talking is started, every thing working fine with DIAL-PLAN and DB look up.
Now, i want to do billing on customer dialed my DID, and from that actually
it DIALED 5 numbers, how can i DO real time billing
into this situation, like numbers can be different It can be ISD,STD,Local
and also free .
if customer having initial balance of $100 then how can i check balance
every time.in a situation once balance is nil then i want to disconnect
calls . is any one facing this type of situation.
give me some idea ,
regards
Dhaval
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Dhaval,
This sounds very much like a system I'm working on for a client right now.
I'm not permitted to disclose much about it due to the NDA i signed, but
I'll risk giving you a point in the right direction.
First, you should create a table in your database that has a column called
callid, and other columns that you will have to decide upon. This table will
be called something like 'call_references'. Oh, and you'll want to define
callid as the primary key for records in that table, but DO NOT make it an
autoincrement, you're going to populate it with a value that is described in
the next step.
Second, at the beginning of the original call you mentioned, define a
variable that will be unique to that call. I personally have done this by
stripping all non-digits from the caller's callerid (using
Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.
Next (this is where I have to start being a bit vague), you're going to
perform an INSERT query, creating a new call_references record (using that
variable I just showed you how to construct as callid's value).
Now, when you defined that variable, you should have preceded the variable
name with two underscores ( __ ), which will tell Asterisk that channels
spawned by the current channel will inherit that variable and it's value.
Voila, you now have a method for storing realtime data such as billing
information between MULTIPLE calls.
I wish I could tell you more, but I can't violate my client's Non-Disclosure
Agreement.
Hope this helps you out!
Sherwood McGowan
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Re: [asterisk-users] Asterisk Realtime Billing Question???
Tarek,
I'm not sure why it would be our problem is someone came into your
office and started making long distance calls over a trunk I was
providing your company I'm pretty sure that if I had tried that
with some of my carriers in the past they would have laughed until
they cried...
Oh, and also, since this was a wholesale carrier, the customers were
in control of their own freeze amount. It was there to allow THEM to
control their account better. I'd be willing to bet that my clients
would have been happy to just keep billing them for every minute they
used.
Lastly, I would like to just say, I'm not the guy who requested the
feature, I'm the guy who figured out how to make it happen, and making
it happen back in early 2006, when the MySQL addon was just BARELY
stable...
It's ok, I don't need respect, I have the knowledge that I'm the mick,
and I'm awesome :P
Cheers :D
On Thu, Oct 21, 2010 at 4:37 AM, Tarek Sawah tareksa...@hotmail.com wrote:
If you look at it the way you want it.. you usually tell your customer the
available funds and minutes in their account right?
How will you explain politely that you have dropped their calls for lack
of balance because someone else used their account?
If you don't tell them their balance and call duration before call .. then
that won't be a problem.
Now you can do some kind of script to do the math and disconnect calls when
balance is over.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, October 21, 2010 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Realtime Billing Question???
Hi Sherwood ,
well , i think you did not understand my question , i want real time billing
like as i mentioned that if i want to dial 5 number with different call rate
how can i access same
balance into those 5 people, if all are connected how can i periodically
update billing , as you suggested it will assign total balance to those 5
people but actually we can not do like this as total balance of user $100 ,
as per your suggestion it will give $100 for those 5 people which is
practically wrong i think.
give your thougts.
regards
dhaval
On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Hello All,
after so long time i posted a new question regarding billing, hope anyone
have some solution.
I have situation in that i want to do billing of more than 1 call in real
time below are scenario and explanation.
Scenario:
A customer called my DID number and after that from here i dial few number
let say 5 number. once number are placed into DIAL
i will put this customer into conference [MEETME] , once a Members are
picked up call they will also patched into conference and
talking is started, every thing working fine with DIAL-PLAN and DB look up.
Now, i want to do billing on customer dialed my DID, and from that actually
it DIALED 5 numbers, how can i DO real time billing
into this situation, like numbers can be different It can be ISD,STD,Local
and also free .
if customer having initial balance of $100 then how can i check balance
every time.in a situation once balance is nil then i want to disconnect
calls . is any one facing this type of situation.
give me some idea ,
regards
Dhaval
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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Dhaval,
This sounds very much like a system I'm working on for a client right now.
I'm not permitted to disclose much about it due to the NDA i signed, but
I'll risk giving you a point in the right direction.
First, you should create a table in your database that has a column called
callid, and other columns that you will have to decide upon. This table will
be called something like 'call_references'. Oh, and you'll want to define
callid as the primary key for records in that table, but DO NOT make it an
autoincrement, you're going to populate it with a value that is described in
the next step.
Second, at the beginning of the original call you mentioned, define a
variable that will be unique to that call. I personally have done this by
stripping all non-digits from the caller's callerid (using
Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.
Next (this is where I have to start being a bit vague), you're going to
perform an INSERT query, creating a new call_references record (using that
Re: [asterisk-users] Asterisk Realtime Billing Question???
actually my mail was not meant to be disrespectful. it was an inquiry. i have a billing system and had a few of those thoughts regarding real time billing. my issue was explaining to a customer that his call disconnected an hour earlier because someone else used his account.. I'm doing retail not wholesale, you may understand my question more clearly now? Regards Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: sherwood.mcgo...@gmail.com Date: Thu, 21 Oct 2010 05:18:17 -0500 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Realtime Billing Question??? Tarek, I'm not sure why it would be our problem is someone came into your office and started making long distance calls over a trunk I was providing your company I'm pretty sure that if I had tried that with some of my carriers in the past they would have laughed until they cried... Oh, and also, since this was a wholesale carrier, the customers were in control of their own freeze amount. It was there to allow THEM to control their account better. I'd be willing to bet that my clients would have been happy to just keep billing them for every minute they used. Lastly, I would like to just say, I'm not the guy who requested the feature, I'm the guy who figured out how to make it happen, and making it happen back in early 2006, when the MySQL addon was just BARELY stable... It's ok, I don't need respect, I have the knowledge that I'm the mick, and I'm awesome :P Cheers :D On Thu, Oct 21, 2010 at 4:37 AM, Tarek Sawah wrote: If you look at it the way you want it.. you usually tell your customer the available funds and minutes in their account right? How will you explain politely that you have dropped their calls for lack of balance because someone else used their account? If you don't tell them their balance and call duration before call .. then that won't be a problem. Now you can do some kind of script to do the math and disconnect calls when balance is over. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Thursday, October 21, 2010 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Realtime Billing Question??? Hi Sherwood , well , i think you did not understand my question , i want real time billing like as i mentioned that if i want to dial 5 number with different call rate how can i access same balance into those 5 people, if all are connected how can i periodically update billing , as you suggested it will assign total balance to those 5 people but actually we can not do like this as total balance of user $100 , as per your suggestion it will give $100 for those 5 people which is practically wrong i think. give your thougts. regards dhaval On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan wrote: On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA wrote: Hello All, after so long time i posted a new question regarding billing, hope anyone have some solution. I have situation in that i want to do billing of more than 1 call in real time below are scenario and explanation. Scenario: A customer called my DID number and after that from here i dial few number let say 5 number. once number are placed into DIAL i will put this customer into conference [MEETME] , once a Members are picked up call they will also patched into conference and talking is started, every thing working fine with DIAL-PLAN and DB look up. Now, i want to do billing on customer dialed my DID, and from that actually it DIALED 5 numbers, how can i DO real time billing into this situation, like numbers can be different It can be ISD,STD,Local and also free . if customer having initial balance of $100 then how can i check balance every time.in a situation once balance is nil then i want to disconnect calls . is any one facing this type of situation. give me some idea , regards Dhaval -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Dhaval, This sounds very much like a system I'm working on for a client right now. I'm not permitted to disclose much about it due to the NDA i signed, but I'll risk giving you a point in the right direction. First, you should create a table in your database that has a column called callid, and other columns that you will have to decide upon. This
Re: [asterisk-users] Recommendation for a new server
No transcoding? OK, this will work... http://www.supermicro.com/products/system/1U/5015/SYS-5015A-PHF.cfm?typ=E ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Wed, Oct 20, 2010 at 10:52 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, What servers would you suggest for:100 concurrent SIP calls, 4xT1 card, and a not much busy website, i.e. getting 500-1000 hits a day. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendation for a new server
I think I'll prefer Dell over supermicro, as another customer I worked for always complained about supermicro. I also once used supermicro and I had no luck with it. But which model of Dell is good for this requirement? I don't want to get over powerful server than required for this setup. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-21 6:56 AM, Andrew Latham lath...@gmail.com wrote: No transcoding? OK, this will work... http://www.supermicro.com/products/system/1U/5015/SYS-5015A-PHF.cfm?typ=E ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Wed, Oct 20, 2010 at 10:52 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, W... -- _ -- Bandwidth and Colocatio... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to kill AMI ORIGINATE on-the-fly
That would be really difficult to do, to keep track of all three channel events while they are originating and to hangup the failed ones . Easy solution in asterisk for this would be to originate using Local channel and in dialplan use Dial command to make call to all the operators using '' operator. And dial command automatically hangups rest of the channels if one channel succeeds. http://blog.godson.in/2010/10/asterisk-vs-freeswitch-channel-tracking.html On Thu, Oct 21, 2010 at 6:19 AM, Valter Nogueira vgnogue...@gmail.comwrote: My application fires several calls thru AMI ORIGINATE command. For example if I have 3 operators I do 3 ORIGINATEs. My trouble is when one operator quit for some reason, I should kill the corresponding ORIGINATE. Of course, I could let the call ring and hangup after the customer pick-up. But this is not the case, I do have to kill the corresponding ORIGINATE. I could execute a soft hangup, but I am not aware the channel that ORIGINATE is using until customer either pick-up or not answer - generating a OriginateResponse event. I have tried to send a STATUS command on the ActionId, but the answer is assync and I can't trust on it. I would appreciate any help. Thanks, Valter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Regards, Godson Gera http://godson.in -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIALSTATUS always returns NOANSWER
Maybe you should post this portion for your dialplan. I have done the same thing several times and never had this timeout issue. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-21 4:08 AM, GBR Icasiano, Ryan A. raicasi...@globalbridgeresources.com wrote: Hi, Here is the scenario: 1. 1st phone calls and asterisk dials to extension no. 2. Extension answers 1st caller(which makes it busy). 2. 2nd phone calls and asterisk dials to extension no. 3. 2nd phone hears a BUSY tone, but have to wait for the timeout to expire(in DIAL cmd) before proceeding to the next step in dialplan 4. Get the current DIALSTATUS, but it returns NOANSWER, instead of BUSY the problem is, since the 2nd caller hears a busy tone, it should not wait for the timeout to expire, and proceed immediately in fetching the DIALSTATUS. I also tried this scenario and used DEV_STATE, but it always returns NOT_INUSE I already assigned qualify=yes in my sip configuration but still to no avail. any ideas? regards, RYAN ICASIANO -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DIALSTATUS always returns NOANSWER
Hi, Which asterisk version are you using. try setting call-limit value in sip.conf and see if it makes any difference. On Thu, Oct 21, 2010 at 1:29 PM, GBR Icasiano, Ryan A. raicasi...@globalbridgeresources.com wrote: Hi, Here is the scenario: 1. 1st phone calls and asterisk dials to extension no. 2. Extension answers 1st caller(which makes it busy). 2. 2nd phone calls and asterisk dials to extension no. 3. 2nd phone hears a BUSY tone, but have to wait for the timeout to expire(in DIAL cmd) before proceeding to the next step in dialplan 4. Get the current DIALSTATUS, but it returns NOANSWER, instead of BUSY the problem is, since the 2nd caller hears a busy tone, it should not wait for the timeout to expire, and proceed immediately in fetching the DIALSTATUS. I also tried this scenario and used DEV_STATE, but it always returns NOT_INUSE I already assigned qualify=yes in my sip configuration but still to no avail. any ideas? regards, RYAN ICASIANO -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Regards, Godson Gera FreeSWITCH Asterisk Billing Consultanthttp://blog.godson.in/2010/10/asterisk-vs-freeswitch-channel-tracking.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback in the middle of a call though AMI
Hi, I am not using 1.6 but in 1.2 or 1.4 there is no straight forward way to do this. The workaround i use it to pull the caller into conference and play what ever I want using a agi script connected to the same conference room. On Wed, Oct 20, 2010 at 4:35 PM, Gustavo Garcia Bernardo g...@tid.es wrote: Hi folks, Is it possible (asterisk 1.6) to trigger the playback of an audio file in the middle of a call using the Manager Interface? I’m looking for something like AMI PlayDTMF command but for audio files. Thanks a lot, G. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Regards, Godson Gera Asterisk FreeSWITCH Billing Consultanthttp://blog.godson.in/2010/10/asterisk-vs-freeswitch-channel-tracking.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendation for a new server
There is always one (or more) bad product from every manufacturer that leaves a bad taste in your mouth. Always keep you mind open and search before you hit the order button. For supply chain I like Supermicro. I don't like all their products but I know I can get the right part with one order number from many distributors. IBM has a good supply chain and their parts are actually fairly priced. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Thu, Oct 21, 2010 at 8:35 AM, Zeeshan Zakaria zisha...@gmail.com wrote: I think I'll prefer Dell over supermicro, as another customer I worked for always complained about supermicro. I also once used supermicro and I had no luck with it. But which model of Dell is good for this requirement? I don't want to get over powerful server than required for this setup. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-21 6:56 AM, Andrew Latham lath...@gmail.com wrote: No transcoding? OK, this will work... http://www.supermicro.com/products/system/1U/5015/SYS-5015A-PHF.cfm?typ=E ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Wed, Oct 20, 2010 at 10:52 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, W... -- _ -- Bandwidth and Colocatio... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Busy detection in dialplan - Asterisk 1.6
We have an employee who works from home. We sent her a SIP phone to work as an extension off our Asterisk 1.6 system, but her DSL service is so bad she was dropping calls all the time. It's not just a tuning or QoS issue. Her service is simply unreliable. She had a POTS line installed and I have the dialplan set up so that when her extension is dialed, it calls out over our SIP provider to her 10-digit POTS number. If she is on the phone and her line is busy, I want Asterisk to place the caller into her Asterisk voicemail rather than hearing a busy signal. The way I have this working currently is by using Followme without a preceding Dial command. Seems that the Followme app handles the busy properly. The problem is that every call she receives is announced and requires her to press 1 to accept or 2 to reject. I suppose I could modify the Followme code, but I'd rather not. Any ideas are appreciated. Thanks. The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialing from asterisk console?
Hello friends, I'm trying to make a simple call from asterisk CLI, but is quite confuse i followed the information here: http://www.voip-info.org/wiki/view/Asterisk+CLI+dial and changed my extensions.conf like this: alsa.conf [general] autoanswer=no context=consolecontext extension=100 By the way, how do i know if my console is using the channel driver ALSA or OSS? then, in extensions.conf: [consolecontext] exten = 100,1,Dial($DEMO) And then, from the Asterisk CLI: many attempts: CLI dial No such extension 's' in context 'default' CLI dial 100 No such extension '100' in context 'default' What am i doing wrong? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendation for a new server
I just put in an HP DL360 G6 for a client spec with a Sangoma 4x PRI, a Sangoma 4x FXS and about 150 devices. Running live now on 1x PRI approx 20-calls and 60 phones, load is at zero. We went with the base machine Xeon 5500 + 4GB RAM, 2x PS, 2x HD (raid mirror)... about 3K$ but Im certain we are overspec'ed as we are not transcoding and the echo cancel is on the Sangoma board. Client has a rack full of HP, so they preferred to stay branded and be safe when it comes to the specs. Only hiccup I had with rackmounts and FXS cards is that the Sangoma boards need a molex connector which are nowhere to be found on the HPs due to the proprietary power connectors. We ended up cutting out the PCI card's backplate and running a molex-only power cable right into the UPS. Besides the price, the HP machines are pretty swell. Seb On 2010-10-21, at 7:49 AM, Andrew Latham wrote: There is always one (or more) bad product from every manufacturer that leaves a bad taste in your mouth. Always keep you mind open and search before you hit the order button. For supply chain I like Supermicro. I don't like all their products but I know I can get the right part with one order number from many distributors. IBM has a good supply chain and their parts are actually fairly priced. ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Thu, Oct 21, 2010 at 8:35 AM, Zeeshan Zakaria zisha...@gmail.com wrote: I think I'll prefer Dell over supermicro, as another customer I worked for always complained about supermicro. I also once used supermicro and I had no luck with it. But which model of Dell is good for this requirement? I don't want to get over powerful server than required for this setup. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-21 6:56 AM, Andrew Latham lath...@gmail.com wrote: No transcoding? OK, this will work... http://www.supermicro.com/products/system/1U/5015/SYS-5015A-PHF.cfm?typ=E ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Wed, Oct 20, 2010 at 10:52 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, W... -- _ -- Bandwidth and Colocatio... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialing from asterisk console?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias Sent: Thursday, October 21, 2010 7:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] dialing from asterisk console? Hello friends, I'm trying to make a simple call from asterisk CLI, but is quite confuse i followed the information here: http://www.voip-info.org/wiki/view/Asterisk+CLI+dial and changed my extensions.conf like this: alsa.conf [general] autoanswer=no context=consolecontext extension=100 By the way, how do i know if my console is using the channel driver ALSA or OSS? then, in extensions.conf: [consolecontext] exten = 100,1,Dial($DEMO) And then, from the Asterisk CLI: many attempts: CLI dial No such extension 's' in context 'default' CLI dial 100 No such extension '100' in context 'default' What am i doing wrong? Dial from CLI works differently from regular Dial in the Dialplan. From what I can see, you are limited (1.4.30) to dialing extensions defined in the [default] context . Do dialplan show default and you'll see what you can dial. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dialing from asterisk console?
Try: dial 1...@consolecontext -Original Message- From: Danny Dias ing.diasda...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 21 Oct 2010 14:41:47 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] dialing from asterisk console? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.8 SIP register uri: peer field ?
Hello, Looking the asterisk 1.8 API documentation (http://www.asterisk.org/astdocs/api/index.html), I see a lot of new fields for sip register uris: register = [peer?][transport://]us...@domain][:secret[:authuse...@host[:port][/extension][~expiry] But the *peer* is not explained anywhere. What it is for ? Regards, Guillaume Bour. -- Guillaume Bourgb...@proformatique.com - proformatique 10 bis, rue Lucien VOILIN - 92800 Puteaux -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Thursday, October 21, 2010 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Busy detection in dialplan - Asterisk 1.6 We have an employee who works from home. We sent her a SIP phone to work as an extension off our Asterisk 1.6 system, but her DSL service is so bad she was dropping calls all the time. It's not just a tuning or QoS issue. Her service is simply unreliable. She had a POTS line installed and I have the dialplan set up so that when her extension is dialed, it calls out over our SIP provider to her 10-digit POTS number. If she is on the phone and her line is busy, I want Asterisk to place the caller into her Asterisk voicemail rather than hearing a busy signal. The way I have this working currently is by using Followme without a preceding Dial command. Seems that the Followme app handles the busy properly. The problem is that every call she receives is announced and requires her to press 1 to accept or 2 to reject. I suppose I could modify the Followme code, but I'd rather not. Any ideas are appreciated. Thanks. I know how this works with DAHDI/POTS; don't know what it will do dialing over SIP Exten = 1234,1,Dial(DAHDI/1/w5551212,20,KkTt) Exten = 1234,n,voicemail(1...@default) Exten = 1234,n,hangup Exten = 1234-BUSY,1,voicemail(1...@default) Exten = 1234-CONGESTION,1,voicemail(1...@default) When I dial 1234, the other side has 20 seconds (about 4 rings) to pick up. If no pickup, voicemail is called. Lines 4 and 5 might (or might not) be redundant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error with Connecting Two Asterisk BOX with IAX
Yhank you very much Giorgio, now work with the general option: calltokenoptional=0.0.0.0/0.0.0.0 Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On 10/20/2010 04:07 PM, VoIP Question wrote: Hello again, If I set a peer to use G.711 only, they try to process a sent fax in G.711, but Asterisk doesn't like it: WARNING[4903]: res_fax.c:1709 sendfax_t38_init: Audio FAX not allowed on channel 'SIP/Main-000a' and T.38 negotiation failed; aborting. What can I do to enable it? What you can do is read the documentation. The built-in help for the SendFAX application shows you how to enable audio FAX on channels that support T.38 (where audio FAX mode is normally disabled for reliability reasons). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Hi Bruce, can you show agent login/logoff diaplan? Maybe there is a solution but i have to know how yours agents login/logoff. Regards - Bakko -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On 10/20/2010 11:35 AM, VoIP Question wrote: On Wed, Oct 20, 2010 at 4:25 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: This was fixed in Asterisk 1.6.2.12 and later releases, so if you were running the current version, you wouldn't have experienced this specific problem. This was listed in the ChangeLog for 1.6.2.12, but unfortunately the commit message the developer wrote did not explain why the change was made or what problem it was addressing, so you wouldn't have noticed it. In any case, upgrading to 1.6.2.12 or later will cure this problem. I upgraded to 1.6.2.13 and now we get this error (with a specific destination, to which we occasionally need to send faxes): WARNING[857]: udptl.c:1087 ast_udptl_write: (SIP/XXX): UDPTL asked to send 50 bytes of IFP when far end only prepared to accept 30 bytes; data loss will occur.You may need to override the T38FaxMaxDatagram value for this endpoint in the channel driver configuration. How can we fix it, without risking incompatibility with other end-points? What's a channel driver configuration and where is it? It appears that you need to spend some time learning the basics of Asterisk. In this case, the channel driver is chan_sip, since the channel involved is a SIP channel, and the 'channel driver configuration' is the sip.conf file. It is unfortunate that you have chosen to tackle a very complex task (T.38 interoperability is fraught with problems due to widely varying implementations) as your first experience with Asterisk... there's a lot you'll need to learn to be able to diagnose and troubleshoot problems. Asterisk alone is not 'point and click', and adding T.38 to the mix makes things more complicated. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On 10/20/2010 09:35 AM, VoIP Question wrote: Thank you Kevin, We'll upgrade our server to 1.6.2.12 and try again. Another question: Is there (expect for the admin guide that we didn't succeed to understand the example in) an example somewhere for ReceiveFax full extensions.conf diaplan? We would like to allocate one of the extensions that our SIP provider gives us to a fax storage server or later to email. The ReceiveFAX example in the Fax For Asterisk administrator's guide is very straightforward and easy to follow... if you don't understand it, then you'll need to spend some time learning how the Asterisk dialplan works. I would highly recommend reading the O'Reilly Asterisk book (which you can read online for free)... while it is based on Asterisk 1.4, the dialplan concepts documented in it have not changed much in Asterisk 1.6, and gaining that basic understanding will go a long way towards helping you be able to resolve these issues on your own. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Here is the login for English:
;English-temp LOGIN
exten = 800,1,Answer()
exten = 800,n,AddQueueMember(500|Local/${CALLERID(num)}...@from-internal/n)
exten = 800,n,Set(DEVSTATE(Custom:agenten)=INUSE)
exten = 800,n,Playback(agent-loginok)
exten = 800,n,Hangup()
;English Logout
;All Queues Logout
exten = 802,1,Answer
exten = 802,n,RemoveQueueMember(500|Local/${CALLERID(num)}...@from-internal/n)
exten = 802,n,RemoveQueueMember(499|Local/${CALLERID(num)}...@from-internal/n)
exten = 802,n,Playback(agent-loggedoff)
exten = 802,n,Hangup
The logout logs both English and Spanish (which is just like English for
Login.
Thanks,
Bruce
On Thu, Oct 21, 2010 at 10:01 AM, bakko asannu...@gmail.com wrote:
Hi Bruce,
can you show agent login/logoff diaplan?
Maybe there is a solution but i have to know how yours agents login/logoff.
Regards
- Bakko
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[asterisk-users] Asterisk 1.8.0-rc5: Blind transfer failed, SIP REFER Method
Hi, I setup an asterisk system (version 1.8.0-rc5). While using a SIP only environment I discovered a problem using blind transfer. The phones are SNOM or Aastra and are using the SIP REFER Method. The following is working: User A calls user B, B accepts the call, user A than transfers to user C The following is NOT working: User A calls user B, B accepts the call, user B than transfers to user C The call is terminated by asterisk without any warnings or error message in the CLI. Looking at Events on AMI, I can see in the first case an Event Newchannel with a Channel: AsyncGoto... followed by an Event Masquerade in prior to the Transfer. These events are missing in the second case. Is this a new bug or do I something wrong? Shall I open an issue on the tracker? Thanks for any hints, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.80-rc5
Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician
We have a couple of positions open, please respond to the posting if qualified and interested. http://www.ntegrated.net/resources/job-opportunities/field-service-install-technician http://www.ntegrated.net/resources/job-opportunities/network-engineering-voice These are full time positions in Dallas, no telecommuters please. Thanks. JR -- JR Richardson Engineering for the Masses -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.80-rc5
On Thu, Oct 21, 2010 at 10:40 AM, Dave Cotton dcot...@linuxautrement.com wrote: errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. What OS are you running? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.80-rc5
On Thu, Oct 21, 2010 at 11:05 AM, Paul Belanger paul.belan...@polybeacon.com wrote: What OS are you running? If I had to guess SUSE? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician
On 21 Oct 2010, at 15:56, JR Richardson wrote: These are full time positions in Dallas, no telecommuters please. A very vast majority of people on here are not in Dallas (and indeed probably a majority in the US). So stop filling their mailboxes with this crap. Incase you hadn't noticed Asterisk Users Mailing List - Non-Commercial Discussion S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.80-rc5
On 21/10/10 17:05, Paul Belanger wrote: On Thu, Oct 21, 2010 at 10:40 AM, Dave Cotton dcot...@linuxautrement.com wrote: errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. What OS are you running? Suse 11.3 X86_64 and if there is a required statement in /etc/asterisk/modules.conf it fails with:- [Oct 21 17:09:48] NOTICE[17222]: loader.c:1118 load_modules: 192 modules will be loaded. *** Failed to load module chan_iax2.so - Required [Oct 21 17:09:48] ERROR[17222]: loader.c:958 load_resource_list: *** Failed to load module chan_iax2.so - Required Dave Cotton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.80-rc5
On 21/10/10 16:40, Dave Cotton wrote: More interesting is that after make samples I have no iax2 available. Adding more info :- [Oct 21 17:14:04] WARNING[17255]: loader.c:387 load_dynamic_module: Error loading module 'res_crypto': /usr/lib/asterisk/modules/res_crypto.so: cannot open shared object file: No such file or directory [Oct 21 17:14:04] WARNING[17255]: loader.c:449 load_dynamic_module: Error loading module 'pbx_dundi.so': /usr/lib/asterisk/modules/pbx_dundi.so: undefined symbol: ast_check_signature_bin [Oct 21 17:14:04] WARNING[17255]: loader.c:839 load_resource: Module 'pbx_dundi.so' could not be loaded. [Oct 21 17:14:04] WARNING[17255]: loader.c:387 load_dynamic_module: Error loading module 'res_crypto': /usr/lib/asterisk/modules/res_crypto.so: cannot open shared object file: No such file or directory [Oct 21 17:14:04] WARNING[17255]: loader.c:387 load_dynamic_module: Error loading module 'res_crypto': /usr/lib/asterisk/modules/res_crypto.so: cannot open shared object file: No such file or directory [Oct 21 17:14:04] WARNING[17255]: loader.c:449 load_dynamic_module: Error loading module 'func_aes.so': /usr/lib/asterisk/modules/func_aes.so: undefined symbol: ast_aes_set_decrypt_key [Oct 21 17:14:04] WARNING[17255]: loader.c:839 load_resource: Module 'func_aes.so' could not be loaded. [Oct 21 17:14:04] WARNING[17255]: loader.c:387 load_dynamic_module: Error loading module 'res_crypto': /usr/lib/asterisk/modules/res_crypto.so: cannot open shared object file: No such file or directory [Oct 21 17:14:04] WARNING[17255]: loader.c:449 load_dynamic_module: Error loading module 'chan_iax2.so': /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_aes_set_decrypt_key Dave Cotton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.80-rc5
On Thu, Oct 21, 2010 at 11:12 AM, Dave Cotton dcot...@linuxautrement.com wrote: Suse 11.3 X86_64 Try this patch for the init.d issue: http://asterisk.pastebin.ca/1969072 after you've applied it rerun: $ make configs As for issue 2, I suspect you don't have res_crypto.so built. Can you attach your config.log -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician
His post may have been of interest to some outside of DFW, and I appreciated your post less than his. But, enjoy. C == A very vast majority of people on here are not in Dallas (and indeed probably a majority in the US). So stop filling their mailboxes with this crap. Incase you hadn't noticed Asterisk Users Mailing List - Non-Commercial Discussion -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician
On Oct 21, 2010, at 11:11 AM, Steve Howes wrote: On 21 Oct 2010, at 15:56, JR Richardson wrote: These are full time positions in Dallas, no telecommuters please. A very vast majority of people on here are not in Dallas (and indeed probably a majority in the US). So stop filling their mailboxes with this crap. Incase you hadn't noticed Asterisk Users Mailing List - Non-Commercial Discussion S JR is pretty active here. No reason to be pissy about it. I'd rather have his crap than your tantrum on someone who contributes to the project. ---fred http://qxork.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.80-rc5
On Thu, Oct 21, 2010 at 11:15 AM, Dave Cotton dcot...@linuxautrement.com wrote: Adding more info :- Ya, so that is the issue. chan_iax2 uses res_crypto, and you likely are missing libssl-dev (openssl) on our box. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.80-rc5
On 21/10/10 17:19, Paul Belanger wrote: On Thu, Oct 21, 2010 at 11:15 AM, Dave Cotton dcot...@linuxautrement.com wrote: Adding more info :- Ya, so that is the issue. chan_iax2 uses res_crypto, and you likely are missing libssl-dev (openssl) on our box. Yes and ./configure and make menuselect did not signal it. :( Dave Cotton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1 way audio asterisk 1.6
Hi I wonder if anyone could give some light on SIP NAT. I've having a friken headache with SIP NAT 1 way audio. Client - NAT - NAT - Server Client can hear users from server side but server cant hear client. Ive tried every possible settings externip set localip set NAT= yes / route directmedia yes/ no Ive check the sip headers in the debug mode and its using the external address in both client and server. Ive tried STUn servers etc No luck. any info on this Its for my installation which I am testing. Zakir-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Blacklisting
Hi, Given the recent increase in SIP brute force attacks, I've had a little idea. The standard scripts that block after X attempts work well to prevent you actually being compromised, but once you've been 'found' then the attempts seem to keep coming for quite some time. Older versions of sipvicious don't appear to stop once you start sending un-reachables (or straight drops). Now this isn't a problem for Asterisk, but it does add up in (noticeable) bandwidth costs - and for people running on lower bandwidth connections. The tool to crash sipvicious can help this, but very few attackers seem to obey it.. The only way I can see to alleviate this, is to blacklist hows *before* they attack. This means you wont ever be targeted past an initial scan. Is there any interest in a 'shared' blacklist (similar to spam blacklists, but obviously implemented in a way that is more usable with Asterisk/iptables)?. Clearly it raises issues about false positives etc, but requiring reports from more than X hosts should alleviate this. There's all the usual de-listing / false-listing worries as with any blacklist, but the SMTP world has solutions we could learn from. Leaving a 'honeypot' running on a single IP address has revealed a few hundred addresses in less than a month. I am fairly certain these are all 'bad' as this host isn't used for anything else. There is obviously a wealth of data (and attacks) out there that would be good to share. Anyone have any thoughts? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
Always start here... http://www.spamhaus.org/drop/ If the AS is stolen, you can block the network and never have to worry about it... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Thu, Oct 21, 2010 at 12:41 PM, Steve Howes steve-li...@geekinter.net wrote: Hi, Given the recent increase in SIP brute force attacks, I've had a little idea. The standard scripts that block after X attempts work well to prevent you actually being compromised, but once you've been 'found' then the attempts seem to keep coming for quite some time. Older versions of sipvicious don't appear to stop once you start sending un-reachables (or straight drops). Now this isn't a problem for Asterisk, but it does add up in (noticeable) bandwidth costs - and for people running on lower bandwidth connections. The tool to crash sipvicious can help this, but very few attackers seem to obey it.. The only way I can see to alleviate this, is to blacklist hows *before* they attack. This means you wont ever be targeted past an initial scan. Is there any interest in a 'shared' blacklist (similar to spam blacklists, but obviously implemented in a way that is more usable with Asterisk/iptables)?. Clearly it raises issues about false positives etc, but requiring reports from more than X hosts should alleviate this. There's all the usual de-listing / false-listing worries as with any blacklist, but the SMTP world has solutions we could learn from. Leaving a 'honeypot' running on a single IP address has revealed a few hundred addresses in less than a month. I am fairly certain these are all 'bad' as this host isn't used for anything else. There is obviously a wealth of data (and attacks) out there that would be good to share. Anyone have any thoughts? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
On Thu, 21 Oct 2010, Steve Howes wrote: Hi, Given the recent increase in SIP brute force attacks, I've had a little idea. The standard scripts that block after X attempts work well to prevent you actually being compromised, but once you've been 'found' then the attempts seem to keep coming for quite some time. Older versions of sipvicious don't appear to stop once you start sending un-reachables (or straight drops). Now this isn't a problem for Asterisk, but it does add up in (noticeable) bandwidth costs - and for people running on lower bandwidth connections. The tool to crash sipvicious can help this, but very few attackers seem to obey it.. The only way I can see to alleviate this, is to blacklist hows *before* they attack. This means you wont ever be targeted past an initial scan. Is there any interest in a 'shared' blacklist (similar to spam blacklists, but obviously implemented in a way that is more usable with Asterisk/iptables)?. Clearly it raises issues about false positives etc, but requiring reports from more than X hosts should alleviate this. There's all the usual de-listing / false-listing worries as with any blacklist, but the SMTP world has solutions we could learn from. Leaving a 'honeypot' running on a single IP address has revealed a few hundred addresses in less than a month. I am fairly certain these are all 'bad' as this host isn't used for anything else. There is obviously a wealth of data (and attacks) out there that would be good to share. Anyone have any thoughts? S -- I'll subscribe, that is for sure. What is the best way to dist the blacklist? iptables include file? Or something more integrated to asterisk... just thinking off the top of my head that a module that vetted inbound connections against an external list would be a very cool thing. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] saturation of bandwidth because of HANGUP
Hello, I have a problem of saturation of bandwidth because of HANGUP which sends thousands of times per second for a single call. Furthermore, the timestamp is still the same for this HANGUP. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime Billing Question???
[snipped very confusing top and bottom posting mix] On Thu, 21 Oct 2010, Sherwood McGowan wrote: Dhaval, You're right, I forgot one thing. The frozen table's id column should not be an autoincrement, it should be set by the insert statement, using the original method I decsribed for creating a unique integer from the callerid number and the current EPOCH. That way, you can be sure that multiple concurrent calls that have frozen funds will only retrieve the record they created. (Oh and, once you thaw the frozen funds, delete the appropriate record in the frozen table) I'm not sure why you think this will only work for a single call at a time. Each time a call occurs that is related to an account will cause more money to be frozen from that account, thereby causing future calls to have less available balance and therefore less time for a call limit. This works for ANY number of concurrent calls on an account, and every one of those calls freezes funds based on the rate at which THAT call's amount to freeze was calculated against. EACH call determines IT'S rate, which is then used to determine the amount to freeze from the account ON THAT CALL. Additionally, since the rate is specific to each call, the limiting of the length of THAT call, your issue of limiting is also a non-issue. I also have worked on the logic for this scenario, and I gave up. Our calling card system now locks a balance and forces the account to one simultaneous call at a time. We report the maximum length of a call to the customer just before the ringing starts, and as someone else stated - to cut it off prematurely is very confusing to the customer (and one of the number one complaints against calling cards - if you sell in Florida it could actually get you in serious trouble). The problem with each call freezing a portion of the balance is that no one call has access to the whole balance, and that was determined (in our case) to be unacceptable, and is definitely unacceptable to the calling card customer. But I don't think we are talking about calling cards. I am guessing that Dhaval is trying to create a termination company, and has customers that maintain a balance with him that want to be able to place multiple simultaneous calls. A common problem. We often end up with negative balances with our upstreams for this very reason - we may be near the bottom of our balance and several calls in progress terminate and bring us below zero. I am sure this is what he is trying to avoid, as the industry is full of people that will simply walk away from a negative balance. Dhaval - your wish, I think, is to manage exactly in real time to decrease the balance as the calls progress. In that way all calls in progress would be cutoff simultaneously as the balance hit zero. That kind of scenario would be very complicated with asterisk. Some external program would have to keep track of the balance and the calls currently in progress, and cut them off at the appropriate time. I would be very interested if anyone has attempted this. I envision something that EVERY SECOND deducts from a balance for every call in progress, at the current rate for each call. Not impossible for sure... Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
With CRON or as an init.d you can do many things... http://www.spamhaus.org/faq/answers.lasso?section=DROP%20FAQ#116 ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Thu, Oct 21, 2010 at 12:54 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Thu, 21 Oct 2010, Steve Howes wrote: Hi, Given the recent increase in SIP brute force attacks, I've had a little idea. The standard scripts that block after X attempts work well to prevent you actually being compromised, but once you've been 'found' then the attempts seem to keep coming for quite some time. Older versions of sipvicious don't appear to stop once you start sending un-reachables (or straight drops). Now this isn't a problem for Asterisk, but it does add up in (noticeable) bandwidth costs - and for people running on lower bandwidth connections. The tool to crash sipvicious can help this, but very few attackers seem to obey it.. The only way I can see to alleviate this, is to blacklist hows *before* they attack. This means you wont ever be targeted past an initial scan. Is there any interest in a 'shared' blacklist (similar to spam blacklists, but obviously implemented in a way that is more usable with Asterisk/iptables)?. Clearly it raises issues about false positives etc, but requiring reports from more than X hosts should alleviate this. There's all the usual de-listing / false-listing worries as with any blacklist, but the SMTP world has solutions we could learn from. Leaving a 'honeypot' running on a single IP address has revealed a few hundred addresses in less than a month. I am fairly certain these are all 'bad' as this host isn't used for anything else. There is obviously a wealth of data (and attacks) out there that would be good to share. Anyone have any thoughts? S -- I'll subscribe, that is for sure. What is the best way to dist the blacklist? iptables include file? Or something more integrated to asterisk... just thinking off the top of my head that a module that vetted inbound connections against an external list would be a very cool thing. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
I was thinking on the same lines, i.e. setup a server which will be regularly updated with these bad IP addresses, and anybody looking to block bad IPs will be able to get this list from here. For example when I get mail from Fail2Ban (which I am getting more and more everyday now), a copy would be sent to this server with the updated bad IP address. But the problem is how to make sure that only legitimate users are contributing to this list. Contributors to this list somehow need to verify to an admin that they are not hackers, and this the hard part. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-21 11:46 AM, Steve Howes steve-li...@geekinter.net wrote: Hi, Given the recent increase in SIP brute force attacks, I've had a little idea. The standard scripts that block after X attempts work well to prevent you actually being compromised, but once you've been 'found' then the attempts seem to keep coming for quite some time. Older versions of sipvicious don't appear to stop once you start sending un-reachables (or straight drops). Now this isn't a problem for Asterisk, but it does add up in (noticeable) bandwidth costs - and for people running on lower bandwidth connections. The tool to crash sipvicious can help this, but very few attackers seem to obey it.. The only way I can see to alleviate this, is to blacklist hows *before* they attack. This means you wont ever be targeted past an initial scan. Is there any interest in a 'shared' blacklist (similar to spam blacklists, but obviously implemented in a way that is more usable with Asterisk/iptables)?. Clearly it raises issues about false positives etc, but requiring reports from more than X hosts should alleviate this. There's all the usual de-listing / false-listing worries as with any blacklist, but the SMTP world has solutions we could learn from. Leaving a 'honeypot' running on a single IP address has revealed a few hundred addresses in less than a month. I am fairly certain these are all 'bad' as this host isn't used for anything else. There is obviously a wealth of data (and attacks) out there that would be good to share. Anyone have any thoughts? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue member status - BUSY
On Thu, 2010-10-21 at 08:15 +0800, GBR Icasiano, Ryan A. wrote: anyone? regards, RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of GBR Icasiano, Ryan A. [raicasi...@globalbridgeresources.com] Sent: Wednesday, October 20, 2010 2:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue member status - BUSY Hi, Is there a way to know if a member of a queue is currently engaged on a call? Or if a queue can return a busy status if all members are currently engaged in a call? QUEUESTATUS only returns FULL and TIMEOUT, and the scenario only falls into TIMEOUT, and has to finish the assigned number of seconds into the QUEUE CMD before it falls back to the next routine on the dialplan. Any ideas? People do not really get that a queue is supposed to work that way. The point of having a queue is that you will have more people waiting than agents available to answer calls, if not why have a queue just make a dial group. The way to do what you want would be to use an AGI that gets a list of agents logged into the queue and see their status. The status for a free agent is 1 so if you do not see any agents with status 1 then all agents are busy. You can then set a variable so you can redirect the caller somewhere else. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
On 21 Oct 2010, at 16:54, Jeff LaCoursiere wrote: I'll subscribe, that is for sure. What is the best way to dist the blacklist? iptables include file? Or something more integrated to asterisk... just thinking off the top of my head that a module that vetted inbound connections against an external list would be a very cool thing. I was thinking some sort of script to pull via HTTP to update whatever you wanted (output as iptables etc). I know its not an instant 'lookup', but an hour delay between updates is nothing. Also means whoever is running the server isn't getting hammered by everyone ;) Realtime lookups from Asterisk would be quite a load (and would introduce latency). S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
We would be interested. Spam is a harder problem to fight due to volume and the ability of any idiot to set up free email accounts. But anyone blasting SIP systems is a pure commercial crook. Tagging and strangling them should be a clear cut project. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Thursday, October 21, 2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP Blacklisting Hi, Given the recent increase in SIP brute force attacks, I've had a little idea. The standard scripts that block after X attempts work well to prevent you actually being compromised, but once you've been 'found' then the attempts seem to keep coming for quite some time. Older versions of sipvicious don't appear to stop once you start sending un-reachables (or straight drops). Now this isn't a problem for Asterisk, but it does add up in (noticeable) bandwidth costs - and for people running on lower bandwidth connections. The tool to crash sipvicious can help this, but very few attackers seem to obey it.. The only way I can see to alleviate this, is to blacklist hows *before* they attack. This means you wont ever be targeted past an initial scan. Is there any interest in a 'shared' blacklist (similar to spam blacklists, but obviously implemented in a way that is more usable with Asterisk/iptables)?. Clearly it raises issues about false positives etc, but requiring reports from more than X hosts should alleviate this. There's all the usual de-listing / false-listing worries as with any blacklist, but the SMTP world has solutions we could learn from. Leaving a 'honeypot' running on a single IP address has revealed a few hundred addresses in less than a month. I am fairly certain these are all 'bad' as this host isn't used for anything else. There is obviously a wealth of data (and attacks) out there that would be good to share. Anyone have any thoughts? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime Billing Question???
Tarek, Ouch, I'm quite sorry. I couldn't sleep when I tried to around 4:30AM after working on a project all night. Unfortunately, I'm not quite sure what your question was... :( Maybe when I wake up a bit more On Thu, Oct 21, 2010 at 5:38 AM, Tarek Sawah tareksa...@hotmail.com wrote: actually my mail was not meant to be disrespectful. it was an inquiry. i have a billing system and had a few of those thoughts regarding real time billing. my issue was explaining to a customer that his call disconnected an hour earlier because someone else used his account.. I'm doing retail not wholesale, you may understand my question more clearly now? Regards Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: sherwood.mcgo...@gmail.com Date: Thu, 21 Oct 2010 05:18:17 -0500 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Realtime Billing Question??? Tarek, I'm not sure why it would be our problem is someone came into your office and started making long distance calls over a trunk I was providing your company I'm pretty sure that if I had tried that with some of my carriers in the past they would have laughed until they cried... Oh, and also, since this was a wholesale carrier, the customers were in control of their own freeze amount. It was there to allow THEM to control their account better. I'd be willing to bet that my clients would have been happy to just keep billing them for every minute they used. Lastly, I would like to just say, I'm not the guy who requested the feature, I'm the guy who figured out how to make it happen, and making it happen back in early 2006, when the MySQL addon was just BARELY stable... It's ok, I don't need respect, I have the knowledge that I'm the mick, and I'm awesome :P Cheers :D On Thu, Oct 21, 2010 at 4:37 AM, Tarek Sawah wrote: If you look at it the way you want it.. you usually tell your customer the available funds and minutes in their account right? How will you explain politely that you have dropped their calls for lack of balance because someone else used their account? If you don't tell them their balance and call duration before call .. then that won't be a problem. Now you can do some kind of script to do the math and disconnect calls when balance is over. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Thursday, October 21, 2010 9:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Realtime Billing Question??? Hi Sherwood , well , i think you did not understand my question , i want real time billing like as i mentioned that if i want to dial 5 number with different call rate how can i access same balance into those 5 people, if all are connected how can i periodically update billing , as you suggested it will assign total balance to those 5 people but actually we can not do like this as total balance of user $100 , as per your suggestion it will give $100 for those 5 people which is practically wrong i think. give your thougts. regards dhaval On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan wrote: On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA wrote: Hello All, after so long time i posted a new question regarding billing, hope anyone have some solution. I have situation in that i want to do billing of more than 1 call in real time below are scenario and explanation. Scenario: A customer called my DID number and after that from here i dial few number let say 5 number. once number are placed into DIAL i will put this customer into conference [MEETME] , once a Members are picked up call they will also patched into conference and talking is started, every thing working fine with DIAL-PLAN and DB look up. Now, i want to do billing on customer dialed my DID, and from that actually it DIALED 5 numbers, how can i DO real time billing into this situation, like numbers can be different It can be ISD,STD,Local and also free . if customer having initial balance of $100 then how can i check balance every time.in a situation once balance is nil then i want to disconnect calls . is any one facing this type of situation. give me some idea , regards Dhaval -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Dhaval, This sounds very much like a system I'm
Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6
Didn't work. It correctly times out after 20 seconds and continues to voicemail, but the caller still hears the remote busy signal during those 20 seconds. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, October 21, 2010 9:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins Sent: Thursday, October 21, 2010 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Busy detection in dialplan - Asterisk 1.6 We have an employee who works from home. We sent her a SIP phone to work as an extension off our Asterisk 1.6 system, but her DSL service is so bad she was dropping calls all the time. It's not just a tuning or QoS issue. Her service is simply unreliable. She had a POTS line installed and I have the dialplan set up so that when her extension is dialed, it calls out over our SIP provider to her 10-digit POTS number. If she is on the phone and her line is busy, I want Asterisk to place the caller into her Asterisk voicemail rather than hearing a busy signal. The way I have this working currently is by using Followme without a preceding Dial command. Seems that the Followme app handles the busy properly. The problem is that every call she receives is announced and requires her to press 1 to accept or 2 to reject. I suppose I could modify the Followme code, but I'd rather not. Any ideas are appreciated. Thanks. I know how this works with DAHDI/POTS; don't know what it will do dialing over SIP Exten = 1234,1,Dial(DAHDI/1/w5551212,20,KkTt) Exten = 1234,n,voicemail(1...@default) Exten = 1234,n,hangup Exten = 1234-BUSY,1,voicemail(1...@default) Exten = 1234-CONGESTION,1,voicemail(1...@default) When I dial 1234, the other side has 20 seconds (about 4 rings) to pick up. If no pickup, voicemail is called. Lines 4 and 5 might (or might not) be redundant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
On Thu, 21 Oct 2010, Andrew Latham wrote: Always start here... http://www.spamhaus.org/drop/ If the AS is stolen, you can block the network and never have to worry about it... ~ Andrew lathama Latham lath...@gmail.com I guess you are assuming that spam networks should be included in the blacklist by default? I'm not sure that is a good assumption. Some of my customer netblocks have ended up on spam lists unknowingly (by leaving open SMTP servers for example), and if that had affected their ability to place phone calls also it would have been disastrous. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
On 10/21/10 12:07 PM, Steve Howes steve-li...@geekinter.net wrote: On 21 Oct 2010, at 16:54, Jeff LaCoursiere wrote: I'll subscribe, that is for sure. What is the best way to dist the blacklist? iptables include file? Or something more integrated to asterisk... just thinking off the top of my head that a module that vetted inbound connections against an external list would be a very cool thing. I was thinking some sort of script to pull via HTTP to update whatever you wanted (output as iptables etc). I know its not an instant 'lookup', but an hour delay between updates is nothing. Also means whoever is running the server isn't getting hammered by everyone ;) Realtime lookups from Asterisk would be quite a load (and would introduce latency). I would think DNS would be the best way. Querying it in real shouldn't be a problem and the zone could be replicated to a local server if need be. -Dave -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
On 21 Oct 2010, at 17:03, Zeeshan Zakaria wrote: But the problem is how to make sure that only legitimate users are contributing to this list. Contributors to this list somehow need to verify to an admin that they are not hackers, and this the hard part. I was thinking of having a threshold of number of people reporting an address before it's approved (perhaps from X countries to stop someone with their own subnet abusing it). Clearly it's not an easy thing to guarantee, but a 'report false positive' with human intervention at this point might be useful. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
On Thu, 21 Oct 2010, Steve Howes wrote: On 21 Oct 2010, at 16:54, Jeff LaCoursiere wrote: I'll subscribe, that is for sure. What is the best way to dist the blacklist? iptables include file? Or something more integrated to asterisk... just thinking off the top of my head that a module that vetted inbound connections against an external list would be a very cool thing. I was thinking some sort of script to pull via HTTP to update whatever you wanted (output as iptables etc). I know its not an instant 'lookup', but an hour delay between updates is nothing. Also means whoever is running the server isn't getting hammered by everyone ;) Realtime lookups from Asterisk would be quite a load (and would introduce latency). S -- I agree in principle - some cron job pulling the list by http would certainly be simple. But just to continue my thoughts to the brick wall, I don't see a lookup adding latency to the call other than what should be a very brief addition to the time taken for a call to be accepted. Once accepted you would just continue to accept the packets. How about something DNS based? Load could potentially be distributed that way if a number of people agreed to participate. I'll mull this over a bit more. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
On 21 Oct 2010, at 17:32, Jeff LaCoursiere wrote: I agree in principle - some cron job pulling the list by http would certainly be simple. But just to continue my thoughts to the brick wall, I don't see a lookup adding latency to the call other than what should be a very brief addition to the time taken for a call to be accepted. Yea that's what I was referring to. Say some evil people attacked the server, you could add a few second delay to someone's call setup. I know it's not a major problem but it might just be opening another attack vector. Once accepted you would just continue to accept the packets. How about something DNS based? Load could potentially be distributed that way if a number of people agreed to participate. I'll mull this over a bit more. DNS is a possibility. It would require an Asterisk module I guess. There's nothing saying we could publish the same data in multiple ways (store it in SQL somewhere and output files to HTTP and generated zone files for bind to pick up). S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
Always start here... http://www.spamhaus.org/drop/ If the AS is stolen, you can block the network and never have to worry about it... I guess you are assuming that spam networks should be included in the blacklist by default? I'm not sure that is a good assumption. Some of my customer netblocks have ended up on spam lists unknowingly (by leaving open SMTP servers for example), and if that had affected their ability to place phone calls also it would have been disastrous. j Take TWO minutes and read http://www.spamhaus.org/drop/ . Add some items to your BGP route lists and smile at the decrease in traffic. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.80-rc5
On Thu, Oct 21, 2010 at 11:25 AM, Dave Cotton dcot...@linuxautrement.com wrote: Yes and ./configure and make menuselect did not signal it. :( Did the patch at-least work for you? -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hardware Compatibility HP Proliant - Sangoma PCI Express
Hi to all, I am in the process of setup a new asterisk server, I think in the HP Proliant ML350 G6 Server (aprox. 100 SIP Users), and Sangoma A102DE Card. The specs of the Proliant (HP PART 487932-001) about PCI are the next. 1 ( 1 ) x PCI Express 2.0 x16 ( x8 mode ) , 1 ( 1 ) x PCI Express 2.0 x8 ( x8 mode ) , 4 ( 3 ) x PCI Express 2.0 x8 ( x4 mode ) The question is, if the card is compatible with the PCI slots in the server? And. If there is a known issue with this combination? Thanks a lot. Ricardo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.80-rc5
On 21/10/10 19:26, Paul Belanger wrote: On Thu, Oct 21, 2010 at 11:25 AM, Dave Cotton dcot...@linuxautrement.com wrote: Yes and ./configure and make menuselect did not signal it. :( Did the patch at-least work for you? I'd already edited the init file so I didn't use it.. Dave Cotton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.0 Now Available!
The Asterisk Development Team is proud to announce the release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page. http://www.asterisk.org/asterisk-versions The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we've had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release. You can find a summary of the work involved with the 1.8.0 release in the sumary: http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt A short list of available features includes: * Secure RTP * IPv6 Support in the SIP channel driver * Connected Party Identification Support * Calendaring Integration * A new call logging system, Channel Event Logging (CEL) * Distributed Device State using Jabber/XMPP PubSub * Call Completion Supplementary Services support * Advice of Charge support * Much, much more! A full list of new features can be found in the CHANGES file. http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup For a full list of changes in the current release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Why high latency on internal lan?
I have a 100MB internal lan. aastra's are wired. asterisk box is wired next to the switch. But look: sip show peers 142/14210.10.10.42 D A 5060 OK (137 ms) 144/14410.10.10.44 D A 5060 OK (136 ms) 145/14510.10.10.45 D A 5060 OK (168 ms) 150/15010.10.10.50 D A 5060 OK (136 ms) 152/15210.10.10.52 D A 5060 OK (133 ms) 153/15310.10.10.53 D A 5060 OK (135 ms) ping from the asterisk box to any of the internal phones is 1ms. My traceroute [v0.75] PBX Thu Oct 21 13:58:36 2010 Host 10.10.10.42 Loss% Snt Last Avg Best Wrst StDev 0.0%650.6 0.6 0.5 1.3 0.2 And latencies to outside sip providers are low: jnctn/... 66.227.100.20 5060 OK (7 ms) teliax/..8.14.120.23 N 5060 OK (7 ms) What gives? Isn't sip show peers measuring latency? Why so different from ping? And, more importantly, do I have a problem? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware Compatibility HP Proliant - Sangoma PCI Express
Am 21.10.2010 19:30, schrieb Ricardo Melendez: Hi to all, I am in the process of setup a new asterisk server, I think in the HP Proliant ML350 G6 Server (aprox. 100 SIP Users), and Sangoma A102DE Card. The specs of the Proliant (HP PART 487932-001) about PCI are the next. 1 ( 1 ) x PCI Express 2.0 x16 ( x8 mode ) , 1 ( 1 ) x PCI Express 2.0 x8 ( x8 mode ) , 4 ( 3 ) x PCI Express 2.0 x8 ( x4 mode ) The question is, if the card is compatible with the PCI slots in the server? And. If there is a known issue with this combination? Thanks a lot. Ricardo Hello, i dont have a ML360 but several DL380 G5 with Sangoma A108D cards in it and i dont have any problems with this even if all 240 channels are used. best regards stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why high latency on internal lan?
Am 21.10.2010 20:03, schrieb sean darcy: I have a 100MB internal lan. aastra's are wired. asterisk box is wired next to the switch. But look: sip show peers 142/14210.10.10.42 D A 5060 OK (137 ms) 144/14410.10.10.44 D A 5060 OK (136 ms) 145/14510.10.10.45 D A 5060 OK (168 ms) 150/15010.10.10.50 D A 5060 OK (136 ms) 152/15210.10.10.52 D A 5060 OK (133 ms) 153/15310.10.10.53 D A 5060 OK (135 ms) ping from the asterisk box to any of the internal phones is 1ms. My traceroute [v0.75] PBX Thu Oct 21 13:58:36 2010 Host 10.10.10.42 Loss% Snt Last Avg Best Wrst StDev 0.0%650.6 0.6 0.5 1.3 0.2 And latencies to outside sip providers are low: jnctn/... 66.227.100.20 5060 OK (7 ms) teliax/..8.14.120.23 N 5060 OK (7 ms) What gives? Isn't sip show peers measuring latency? Why so different from ping? And, more importantly, do I have a problem? sean hello sean, the value you see with sip show peers is the latency of an sip option message from asterisk to your peer. this has nothing to do with ICMP messages like ping. the reason for this could be high sip traffic on your asterisk server. or maybe some locks in the background which needs long to be solved until the incoming package could be processed. you didnt say which version you use so its hard to guess. Even if all these phones are in use they could response slower to an option message than if they are idle. best regards stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming calls
Hi all, After a lot of trouble with a TE110p working with mfcr2 , brazil variant, everything looks great,but I can not go out of my calls. When I try I receive the following log: == Using SIP RTP CoS mark 5-- Executing [33220...@local:1] Dial(SIP/4804-001a, DAHDI/g11/33220567,,T) in new stack == Everyone is busy/congested at this time (1:0/1/0)-- Auto fallthrough, channel 'SIP/4804-001a' status is 'CONGESTION'This is my dahdi show status:Digium Wildcard TE110P T1/E1 Card 0 REC 0 0 0 CAS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)thi´s my dahdi show channels: asterisk*CLI dahdi show channels Chan Extension Context Language MOH InterpretBlockedState pseudodefault default In Service 1 4800 default default In Service 2 4800 defaultdefault In Service 3 4805 defaultdefault In Service 4defaultdefault In Service 5defaultdefault In Service 6defaultdefault In Service 7default default In Service 8default default In Service 9default default In Service 10 defaultdefault In Service 11 defaultdefault In Service 12defaultdefault In Service 13defaultdefault In Service 14defaultdefault In Service 15defaultdefault In Service 17default default In Service 18default default In Service 19default default In Service 20 defaultdefault In Service 21 defaultdefault In Service 22defaultdefault In Service 23defaultdefault In Service 24defaultdefault In Service 25defaultdefault In Service 26default default In Service 27default default In Service 28default default In Service 29 defaultdefault In Service 30 defaultdefault In Service 31defaultdefault In Service *In my incoming call , the log is: MFC/R2 call offered on chan 1. ANI = 1221341400, DNIS = 4804, Category = National SubscriberNew MFC/R2 call detected on chan 2. and don't ring nowhere! Thanks for help!Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.80-rc5
On Thu, 2010-10-21 at 17:12 +0200, Dave Cotton wrote: On 21/10/10 17:05, Paul Belanger wrote: On Thu, Oct 21, 2010 at 10:40 AM, Dave Cotton dcot...@linuxautrement.com wrote: errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. What OS are you running? Suse 11.3 X86_64 For suse there is a precompiled version on the OBS (vitsoft) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6
This did the trick! Masks the busy signal. Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, October 21, 2010 1:22 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Busy detection in dialplan - Asterisk 1.6 Try changing KkTt to rKkTt. This should generate a phony ring until the call is picked up or stops. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this transmission may contain privileged and confidential information. It is intended only for the use of the person(s) named above. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Hi Bruce, with this configuration you can`t control the state of agent. Sorry Regards-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: Incoming calls
Hi, After some changes, the status now is: == Using SIP RTP CoS mark 5-- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,T) in new stack == Everyone is busy/congested at this time (1:0/0/1)-- Auto fallthrough, channel 'SIP/4804-' status is 'CHANUNAVAIL'MFC/R2 call disconnected on channel 1 ThanksAtt, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Subject: Incoming calls Date: Thu, 21 Oct 2010 17:59:35 -0200 Hi all, After a lot of trouble with a TE110p working with mfcr2 , brazil variant, everything looks great,but I can not go out of my calls. When I try I receive the following log: == Using SIP RTP CoS mark 5-- Executing [33220...@local:1] Dial(SIP/4804-001a, DAHDI/g11/33220567,,T) in new stack == Everyone is busy/congested at this time (1:0/1/0)-- Auto fallthrough, channel 'SIP/4804-001a' status is 'CONGESTION'This is my dahdi show status:Digium Wildcard TE110P T1/E1 Card 0 REC 0 0 0 CAS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)thi´s my dahdi show channels: asterisk*CLI dahdi show channels Chan Extension Context Language MOH InterpretBlockedState pseudodefault default In Service 1 4800 default default In Service 2 4800 defaultdefault In Service 3 4805 defaultdefault In Service 4defaultdefault In Service 5defaultdefault In Service 6defaultdefault In Service 7default default In Service 8default default In Service 9default default In Service 10 defaultdefault In Service 11 defaultdefault In Service 12defaultdefault In Service 13defaultdefault In Service 14defaultdefault In Service 15defaultdefault In Service 17default default In Service 18default default In Service 19default default In Service 20 defaultdefault In Service 21 defaultdefault In Service 22defaultdefault In Service 23defaultdefault In Service 24defaultdefault In Service 25defaultdefault In Service 26default default In Service 27default default In Service 28default default In Service 29 defaultdefault In Service 30 defaultdefault In Service 31defaultdefault In Service *In my incoming call , the log is: MFC/R2 call offered on chan 1. ANI = 1221341400, DNIS = 4804, Category = National SubscriberNew MFC/R2 call detected on chan 2. and don't ring nowhere! Thanks for help!Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update
Re: [asterisk-users] Email from Dialplan
I use the following:
Exten = s,n(status-NOTIFY),System(echo '${DIALSTATUS} on
${CALLERID(num)}' at ${STRFTIME(${EPOCH},,%H%M%S)} | mail -s Call
Unsuccessful on DNIS '${ARG10}' neeraj.ch...@ocis.com.au)
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Re: [asterisk-users] MS-SQL / Freetds -- func_odbc
Hi folks, How would I go about running a stored procedure call from asterisk via func_odbc. I'm after an example entry in func_odbc if possible for ast 1.4 Also, if someone could post an insert statement that actually works, would be nice. Thanks, :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue member status - BUSY
Hi, I have modified the way agents are being treated since they are using mobile phones. Having that kind of scenario, it is not recommended to make the agent logged in by using that scenario. Instead, they will call a certain number, login by using the given parameters(company id, username, password) and tag them in the DB as logged in, and their number will ring once a client/customer calls and falls on the queue. Now once asterisk falls to a certain queue, it will then check all members that contains login status on a certain table, then add/delete them in queue_members table in realtime depending on its current login status. This way, it will only ring all currently logged in members. It works fine this way, the only problem is that whenever all members are engaged on a call, their phone is off, etc... the queue cannot determine whether any of them is available or not, as far as I know. regards, RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez [cur...@telecomabmex.com] Sent: Friday, October 22, 2010 12:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue member status - BUSY On Thu, 2010-10-21 at 08:15 +0800, GBR Icasiano, Ryan A. wrote: anyone? regards, RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of GBR Icasiano, Ryan A. [raicasi...@globalbridgeresources.com] Sent: Wednesday, October 20, 2010 2:02 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue member status - BUSY Hi, Is there a way to know if a member of a queue is currently engaged on a call? Or if a queue can return a busy status if all members are currently engaged in a call? QUEUESTATUS only returns FULL and TIMEOUT, and the scenario only falls into TIMEOUT, and has to finish the assigned number of seconds into the QUEUE CMD before it falls back to the next routine on the dialplan. Any ideas? People do not really get that a queue is supposed to work that way. The point of having a queue is that you will have more people waiting than agents available to answer calls, if not why have a queue just make a dial group. The way to do what you want would be to use an AGI that gets a list of agents logged into the queue and see their status. The status for a free agent is 1 so if you do not see any agents with status 1 then all agents are busy. You can then set a variable so you can redirect the caller somewhere else. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Counterpath Presence Patent and Android VoIP app
Is it already Friday? This week Counterpath has two big stories. Todd Carrothers, VP Product Management and Mike Doyle, VP Technology will be on board to tell us more about these two developments and to answer your questions on VUC at 12 noon EDT. 1) Counterpath was granted a patent (# 7,809,381,) for presence detection in mobile and fixed broadband networks. 2) As many of you who participate in VUC Fridays, Counterpath launched their beta Android client a few weeks ago. Join us, it should be an interesting discussion and as always, a long after hours session where we talk about anything VoIP-related, telephony, Internet and network technologies, mobile platforms and so much more... http://vuc.me for all info on ways to call and more about VUC Got SIP ? Call in at 12 Noon EDT on 200...@login.zipdx.com which accepts g711 and g722. We prefer the latter. IRC: #vuc on Freenode.net - web IRC: http://vuc.me/irc Hear you there! /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyway to control the LEDs on the Aastra 55i six top buttons? Maybe through Asterisk?
Thanks for the input. By this configuartion you mean by the way I do Add and Remove member from the Queue? Can you please explain by what sort of configuration (what to use instead of Add and Remove queue member) would get this working. I guess I am looking for speedial/BLF on the same key ?!!! Thanks again On Thu, Oct 21, 2010 at 6:36 PM, bakko asannu...@gmail.com wrote: Hi Bruce, with this configuration you can`t control the state of agent. Sorry Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dials a trunk when off hook
How can I let asterisk immediately dials a trunk when off hook? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dials a trunk when off hook
I am not sure that can be done literally by Asterisk because most phones/adapters give dial tone when off hook, but Asterisk doesn't know the phone is off hook until a send button is pushed, several seconds pass after some keys are pressed, or the # button is pressed. However many of the adapters can be set to autodial. I would look for a phone or adapter that has autodial ability. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Baha @ SH Sent: Friday, October 22, 2010 6:11 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] dials a trunk when off hook How can I let asterisk immediately dials a trunk when off hook? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
