[asterisk-users] announcement to be played for attended transfer call

2013-06-11 Thread Deka, Rajib IN MAA SL
Hello List,



I want to play an announcement for attended transfer calls. For example, A 
calls B, B answers the call and transfers (attended) to C - once transfer 
is complete B should hear an announcement saying you call has been 
transferred. Is there any configuration in asterisk to implement this behavior?



I have not used asterisk Transfer Dialplan application or feature.conf for 
configuring the transfer; however I am using SIP REFER from UA to request the 
transfer.



Regards,

Rajib





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[asterisk-users] how send calls to gatekeeper?

2013-06-11 Thread s m
hello everyone
i have a simple question: i have an asterisk which is a h323 gateway
and has a h323 connection to a cisco gatekeeper and a sip connection
to a pbx.

my question is: how can i send all calls to gatekeeper?

 i searched a lot and found that i should set gatekeeper=192.168.0.X
(ip address of my gatekeeper) in h323.conf file.
but what about extensions.conf file? should i define an extension like
a simple h323 connection to gatekeeper (like
exten=_2.,1,Dial(H323/${EXTEN}@cisco_out,60,))? or no dial pattern
need to be defined in extension.conf file?  if we should define dial
pattern, what is different between a simple trunk h323 connection and
a gateway-gatekeeper h323 connection?

thanks in advance
SAM

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Re: [asterisk-users] announcement to be played for attended transfer call

2013-06-11 Thread jg
While B is talking to C, A is enjoying MOH. You could install a 
musicclass that starts with Your are being


Playing an announcement like Your call has been... to A after C has 
accepted the call is probably not a good idea, because C has to wait 
until the the announcement has finished. In environments where callers 
are announced to C, C would typically not want to wait for A---believe me.


jg

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[asterisk-users] A problem with IAX2

2013-06-11 Thread Mordechay Kaganer
B.H.

Hello!

We have several Asterik boxes that are connected to PSTN using PRI cards
and they are interconnected using IAX2 trunks so that incoming calls are
delivered from PSTN to the servers they belong to.

In past we were using asterisk 1.4 on the server that is receiving IAX
connections and everything worked as expected. Recently, we have switched
to a newer box with asterisk 1.8.22 and then we began to experience
sometimes a strange problem:

At some point of time, incoming IAX connections begin to get refused by the
server and we get the following messages in the logs:

WARNING[] chan_iax2.c: Too much delay in IAX2 calltoken timestamp from
address X.X.X.X

where X.X.X.X is the IP of the PSTN-IAX gateways and all the incoming
calls start to be rejected.

Direct PSTN calls (both incoming and outgoing) to the same server work OK.
The only solution that helps is to kill the asterisk and restart it.

All the servers are connected to the same LAN segment, with gigabit switch,
there is no problems with the network. No packet loss.

There's already bug report present with very similar issue, but it is
suspended and, like stated there, the problem is very hard to reproduce.

See: https://issues.asterisk.org/jira/browse/ASTERISK-21762


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Re: [asterisk-users] A problem with IAX2

2013-06-11 Thread Doug Lytle
 WARNING[] chan_iax2.c: Too much delay in IAX2 calltoken timestamp from 
 address X.X.X.X 

I don't know if this will help, but I have: 

requirecalltoken=no 

In my iax.conf 

Doug 
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Re: [asterisk-users] A problem with IAX2

2013-06-11 Thread Mordechay Kaganer
B.H.

On Tue, Jun 11, 2013 at 3:45 PM, Doug Lytle supp...@drdos.info wrote:

  WARNING[] chan_iax2.c: Too much delay in IAX2 calltoken timestamp
 from address X.X.X.X

 I don't know if this will help, but I have:

 requirecalltoken=no

 In my iax.conf

 Doug


Thanks, Doug. I too have it there and this does not help :-(

Maybe, it's possible to disable calltokens at the originating end?
requirecalltoken=no only tells the receiver to accept calls without a valid
calltoken.



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Re: [asterisk-users] Is uniqueid/sequence a safe CDR table primary key ?

2013-06-11 Thread Jairo
Hello,

Still about CDR and MySQL table, should the calldate field be inserted by
Asterisk?

This is the table structure we are using, based on Asterisk wiki:

mysql describe cdr;
+-+---+--+-+-++
| Field   | Type  | Null | Key | Default |
Extra  |
+-+---+--+-+-++
| id  | mediumint(8) unsigned | NO   | PRI | NULL|
auto_increment |
| calldate| datetime  | NO   | | -00-00 00:00:00
||
| clid| varchar(80)   | NO   | |
||
| src | varchar(80)   | NO   | |
||
| dst | varchar(80)   | NO   | |
||
| dcontext| varchar(80)   | NO   | |
||
| channel | varchar(80)   | NO   | |
||
| dstchannel  | varchar(80)   | NO   | |
||
| lastapp | varchar(80)   | NO   | |
||
| lastdata| varchar(80)   | NO   | |
||
| duration| int(11)   | NO   | | 0
||
| billsec | int(11)   | NO   | | 0
||
| disposition | varchar(45)   | NO   | |
||
| amaflags| int(11)   | NO   | | 0
||
| accountcode | varchar(20)   | NO   | |
||
| uniqueid| varchar(32)   | NO   | |
||
| userfield   | varchar(255)  | NO   | |
||
| peeraccount | varchar(20)   | NO   | |
||
| linkedid| varchar(32)   | NO   | |
||
| sequence| int(11)   | NO   | | 0
||
+-+---+--+-+-++
20 rows in set (0.01 sec)

Thank you!



2013/6/4 Olivier oza_4...@yahoo.fr

 OK, then I'll go with linkedid, uniqueid and sequence number.

 Thanks for sharing this on this list


 2013/6/3 Matthew Jordan mjor...@digium.com

 On 06/03/2013 11:20 AM, Olivier wrote:
  Hi,
 
  When dealing with CDR SQL tables, I always added an auto-incremented
  cdr_id key as a primary key, just in case provided uniqueid key went
 wrong.
 
  Now I'm facing a situation where I need to insert into a database's
  table and from the dialplan, a reference to the CDR record which is
  currently processed.
 
  So my questions are:
 
  1. Can uniqueid/sequence (or uniqueid/sequence/calldate) bundle be
  safely used as CDR's table primary key  (ie I cannot have any
  uniqueid/sequence combination from one CDR record to match a past
  uniqueid/sequence combination) ?

 Possibly. Things to keep in mind:

 * You can run into uniqueid collisions across multiple systems if you do
 not specify a system name in asterisk.conf or do not specify a unique
 system name in asterisk.conf.
 * You can run into uniqueid collisions if your system clock goes
 backwards for any reason (the uniqueid for a channel happens to use a
 timestamp for its uniqueness)

 Whether or not this is unique enough will be completely dependent on
 your overall system configuration.

 In general, the recommended combination that *should* uniquely specify a
 CDR (when configured correctly) is linkedid (which should be enabled and
 added to your schema), uniqueid, and sequence number, with the asterisk
 system name specified.

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org



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Jairo Molina Jr∴
http://www.intermol.com.br
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[asterisk-users] Why does it take several seconds to interpret DTMF-input ?

2013-06-11 Thread Jonas Kellens

Hello,

I notice that it takes 4 to 6 seconds between someone pressing a cipher 
and Asterisk continuing inside the dialplan. How come ???


Taken from verbose logfile :

(attempt 1)
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF begin '1' received on 
SIP/SipAgenT01-1eb0
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF begin ignored '1' on 
SIP/SipAgenT01-1eb0
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF end '1' received on 
SIP/SipAgenT01-1eb0, duration 180 ms
[Jun 11 15:29:25] DTMF[18549] channel.c: DTMF end passthrough '1' on 
SIP/SipAgenT01-1eb0


[Jun 11 15:29:30] VERBOSE[18549] pbx.c: [Jun 11 15:29:30]   == CDR 
updated on SIP/SipAgenT01-1eb0
[Jun 11 15:29:30] VERBOSE[18549] pbx.c: [Jun 11 15:29:30] -- 
Executing [1@pbx-routing:1] Set(SIP/SipAgenT01-1eb0, choice=1) 
in new stack
[Jun 11 15:29:30] VERBOSE[18549] pbx.c: [Jun 11 15:29:30] -- 
Executing [1@pbx-routing:2] System(SIP/SipAgenT01-1eb0, echo 
'418','IVR','1','','SipAgenT01-1eb0','$(date +%s)'  
/var/log/asterisk/loggingAST/SipAgenT01-1eb0.csv) in new stack


(attempt 2)
[Jun 11 15:30:21] DTMF[18780] channel.c: DTMF begin '8' received on 
SIP/SipAgenT01-1ec1
[Jun 11 15:30:21] DTMF[18780] channel.c: DTMF begin ignored '8' on 
SIP/SipAgenT01-1ec1
[Jun 11 15:30:21] DTMF[18780] channel.c: DTMF end '8' received on 
SIP/SipAgenT01-1ec1, duration 160 ms
[Jun 11 15:30:21] DTMF[18780] channel.c: DTMF end passthrough '8' on 
SIP/SipAgenT01-1ec1


[Jun 11 15:30:27] VERBOSE[18780] pbx.c: [Jun 11 15:30:27]   == CDR 
updated on SIP/SipAgenT01-1ec1
[Jun 11 15:30:27] VERBOSE[18780] pbx.c: [Jun 11 15:30:27] -- 
Executing [8@pbx-routing:1] Set(SIP/SipAgenT01-1ec1, choice=8) 
in new stack
[Jun 11 15:30:27] VERBOSE[18780] pbx.c: [Jun 11 15:30:27] -- 
Executing [8@pbx-routing:2] System(SIP/SipAgenT01-1ec1, echo 
'418','IVR','8','','SipAgenT01-1ec1','$(date +%s)'  
/var/log/asterisk/loggingAST/SipAgenT01-1ec1.csv) in new stack




Why doesn't Asterisk continue immediately inside the dialplan after 
having received the DTMF-input ?



Kind regards,

Jonas.
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Re: [asterisk-users] announcement to be played for attended transfer call

2013-06-11 Thread Don Kelly
jg
Sent: Tuesday, June 11, 2013 5:28 AM

Playing an announcement like Your call has been... to A after C has
accepted the call is probably not a good idea, because C has to wait until
the the announcement has finished. In environments where callers are
announced to C, C would typically not want to wait for A---believe me.

**OP asked for an announcement to be played to B, not A.

  --Don



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Re: [asterisk-users] Is uniqueid/sequence a safe CDR table primary key ?

2013-06-11 Thread Kevin Larsen
Are you using cdr_adaptive_odbc.conf to populate it? If so, there is no 
Asterisk analog to calldate. You would need an alias set up. Mine looks 
like:

alias start = calldate

so that the start of my call is what gets logged to the database as the 
calldate.

Kevin Larsen 



From:   Jairo ja...@intermol.com.br
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com, 
Date:   06/11/2013 08:28 AM
Subject:Re: [asterisk-users] Is uniqueid/sequence a safe CDR table 
primary key ?
Sent by:asterisk-users-boun...@lists.digium.com



Hello, 

Still about CDR and MySQL table, should the calldate field be inserted by 
Asterisk?

This is the table structure we are using, based on Asterisk wiki:

mysql describe cdr;
+-+---+--+-+-++
| Field   | Type  | Null | Key | Default | 
Extra  |
+-+---+--+-+-++
| id  | mediumint(8) unsigned | NO   | PRI | NULL| 
auto_increment |
| calldate| datetime  | NO   | | -00-00 00:00:00 
||
| clid| varchar(80)   | NO   | | 
||
| src | varchar(80)   | NO   | | 
||
| dst | varchar(80)   | NO   | | 
||
| dcontext| varchar(80)   | NO   | | 
||
| channel | varchar(80)   | NO   | | 
||
| dstchannel  | varchar(80)   | NO   | | 
||
| lastapp | varchar(80)   | NO   | | 
||
| lastdata| varchar(80)   | NO   | | 
||
| duration| int(11)   | NO   | | 0   
||
| billsec | int(11)   | NO   | | 0   
||
| disposition | varchar(45)   | NO   | | 
||
| amaflags| int(11)   | NO   | | 0   
||
| accountcode | varchar(20)   | NO   | | 
||
| uniqueid| varchar(32)   | NO   | | 
||
| userfield   | varchar(255)  | NO   | | 
||
| peeraccount | varchar(20)   | NO   | | 
||
| linkedid| varchar(32)   | NO   | | 
||
| sequence| int(11)   | NO   | | 0   
||
+-+---+--+-+-++
20 rows in set (0.01 sec)

Thank you!



2013/6/4 Olivier oza_4...@yahoo.fr
OK, then I'll go with linkedid, uniqueid and sequence number.

Thanks for sharing this on this list


2013/6/3 Matthew Jordan mjor...@digium.com
On 06/03/2013 11:20 AM, Olivier wrote:
 Hi,

 When dealing with CDR SQL tables, I always added an auto-incremented
 cdr_id key as a primary key, just in case provided uniqueid key went 
wrong.

 Now I'm facing a situation where I need to insert into a database's
 table and from the dialplan, a reference to the CDR record which is
 currently processed.

 So my questions are:

 1. Can uniqueid/sequence (or uniqueid/sequence/calldate) bundle be
 safely used as CDR's table primary key  (ie I cannot have any
 uniqueid/sequence combination from one CDR record to match a past
 uniqueid/sequence combination) ?

Possibly. Things to keep in mind:

* You can run into uniqueid collisions across multiple systems if you do
not specify a system name in asterisk.conf or do not specify a unique
system name in asterisk.conf.
* You can run into uniqueid collisions if your system clock goes
backwards for any reason (the uniqueid for a channel happens to use a
timestamp for its uniqueness)

Whether or not this is unique enough will be completely dependent on
your overall system configuration.

In general, the recommended combination that *should* uniquely specify a
CDR (when configured correctly) is linkedid (which should be enabled and
added to your schema), uniqueid, and sequence number, with the asterisk
system name specified.

--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org



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Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?

2013-06-11 Thread Matthew J. Roth
Jonas Kellens wrote:
 
 I notice that it takes 4 to 6 seconds between someone pressing a cipher and
 Asterisk continuing inside the dialplan. How come ???
 
 ...
 
 Why doesn't Asterisk continue immediately inside the dialplan after having
 received the DTMF-input ?


Jonas,

Please provide the version of Asterisk you are using and the part of the 
dialplan
that receives the DTMF input.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] A problem with IAX2

2013-06-11 Thread Steve Totaro
On Tue, Jun 11, 2013 at 8:32 AM, Mordechay Kaganer mkaga...@gmail.comwrote:

 B.H.

 Hello!

 We have several Asterik boxes that are connected to PSTN using PRI cards
 and they are interconnected using IAX2 trunks so that incoming calls are
 delivered from PSTN to the servers they belong to.

 In past we were using asterisk 1.4 on the server that is receiving IAX
 connections and everything worked as expected. Recently, we have switched
 to a newer box with asterisk 1.8.22 and then we began to experience
 sometimes a strange problem:

 At some point of time, incoming IAX connections begin to get refused by
 the server and we get the following messages in the logs:

 WARNING[] chan_iax2.c: Too much delay in IAX2 calltoken timestamp from
 address X.X.X.X

 where X.X.X.X is the IP of the PSTN-IAX gateways and all the incoming
 calls start to be rejected.

 Direct PSTN calls (both incoming and outgoing) to the same server work OK.
 The only solution that helps is to kill the asterisk and restart it.

 All the servers are connected to the same LAN segment, with gigabit
 switch, there is no problems with the network. No packet loss.

 There's already bug report present with very similar issue, but it is
 suspended and, like stated there, the problem is very hard to reproduce.

 See: https://issues.asterisk.org/jira/browse/ASTERISK-21762


 --
 משיח NOW!


Use SIP and never look back.

Thanks,
Steve Totaro
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Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?

2013-06-11 Thread Jonas Kellens

On 06/11/2013 04:12 PM, Matthew J. Roth wrote:

Jonas Kellens wrote:

I notice that it takes 4 to 6 seconds between someone pressing a cipher and
Asterisk continuing inside the dialplan. How come ???

...

Why doesn't Asterisk continue immediately inside the dialplan after having
received the DTMF-input ?


Jonas,

Please provide the version of Asterisk you are using and the part of the 
dialplan
that receives the DTMF input.

Regards,

Matthew Roth



Hello,

using Asterisk 1.8.12.2.

Dialplan :

exten = ivr,1,NoOp()
exten = 
ivr,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})

exten = ivr,n,NoOp(${BACKGROUNDSTATUS})
exten = ivr,n,WaitExten(15)
exten = ivr,n,GoTo(restartprompt)

exten = _X,1,Set(choice=${EXTEN})
exten = _X,n,System(echo 
'${klantID}','IVR','${choice}','','${CHANNEL:4}','$(date +%s)'  
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)

exten = _X,n,other_stuff_I_do

exten = _X.,1,Set(choice=${EXTEN})
exten = _X.,n,System(echo 
'${klantID}','IVR','${keuzeID}','','${CHANNEL:4}','$(date +%s)'  
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)

exten = _X.,n,other_stuff_I_do





Kind regards,

Jonas.


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Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?

2013-06-11 Thread Richard Mudgett
On Tue, Jun 11, 2013 at 9:29 AM, Jonas Kellens jonas.kell...@telenet.bewrote:

  On 06/11/2013 04:12 PM, Matthew J. Roth wrote:

 Jonas Kellens wrote:

  I notice that it takes 4 to 6 seconds between someone pressing a cipher and
 Asterisk continuing inside the dialplan. How come ???

 ...

 Why doesn't Asterisk continue immediately inside the dialplan after having

 received the DTMF-input ?

 snip

Dialplan :

 exten = ivr,1,NoOp()
 exten =
 ivr,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
 exten = ivr,n,NoOp(${BACKGROUNDSTATUS})
 exten = ivr,n,WaitExten(15)
 exten = ivr,n,GoTo(restartprompt)

 exten = _X,1,Set(choice=${EXTEN})
 exten = _X,n,System(echo
 '${klantID}','IVR','${choice}','','${CHANNEL:4}','$(date +%s)' 
 /var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
 exten = _X,n,other_stuff_I_do

 exten = _X.,1,Set(choice=${EXTEN})
 exten = _X.,n,System(echo
 '${klantID}','IVR','${keuzeID}','','${CHANNEL:4}','$(date +%s)' 
 /var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
 exten = _X.,n,other_stuff_I_do


It is waiting for more digits because you have asked it for a possible
multi-digit exten and it needs to distinguish between the _X and _X.
patterns.

Richard
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Re: [asterisk-users] Is uniqueid/sequence a safe CDR table primary key ?

2013-06-11 Thread Jairo
Yes, using cdr_adaptive_odbc.conf.

As it is a new table, just changed the name from calldate to start and now
it is inserting the field ok.

Thank you very much for your help.

Best.


2013/6/11 Kevin Larsen kevin.lar...@pioneerballoon.com

 Are you using cdr_adaptive_odbc.conf to populate it? If so, there is no
 Asterisk analog to calldate. You would need an alias set up. Mine looks
 like:

 alias start = calldate

 so that the start of my call is what gets logged to the database as the
 calldate.

 Kevin Larsen



 From:Jairo ja...@intermol.com.br
 To:Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com,
 Date:06/11/2013 08:28 AM
 Subject:Re: [asterisk-users] Is uniqueid/sequence a safe CDR
 table primary key ?
 Sent by:asterisk-users-boun...@lists.digium.com
 --



 Hello,

 Still about CDR and MySQL table, should the calldate field be inserted by
 Asterisk?

 This is the table structure we are using, based on Asterisk wiki:

 mysql describe cdr;

 +-+---+--+-+-++
 | Field   | Type  | Null | Key | Default |
 Extra  |

 +-+---+--+-+-++
 | id  | mediumint(8) unsigned | NO   | PRI | NULL|
 auto_increment |
 | calldate| datetime  | NO   | | -00-00 00:00:00
 ||
 | clid| varchar(80)   | NO   | |
 ||
 | src | varchar(80)   | NO   | |
 ||
 | dst | varchar(80)   | NO   | |
 ||
 | dcontext| varchar(80)   | NO   | |
 ||
 | channel | varchar(80)   | NO   | |
 ||
 | dstchannel  | varchar(80)   | NO   | |
 ||
 | lastapp | varchar(80)   | NO   | |
 ||
 | lastdata| varchar(80)   | NO   | |
 ||
 | duration| int(11)   | NO   | | 0
 ||
 | billsec | int(11)   | NO   | | 0
 ||
 | disposition | varchar(45)   | NO   | |
 ||
 | amaflags| int(11)   | NO   | | 0
 ||
 | accountcode | varchar(20)   | NO   | |
 ||
 | uniqueid| varchar(32)   | NO   | |
 ||
 | userfield   | varchar(255)  | NO   | |
 ||
 | peeraccount | varchar(20)   | NO   | |
 ||
 | linkedid| varchar(32)   | NO   | |
 ||
 | sequence| int(11)   | NO   | | 0
 ||

 +-+---+--+-+-++
 20 rows in set (0.01 sec)

 Thank you!



 2013/6/4 Olivier *oza_4...@yahoo.fr* oza_4...@yahoo.fr
 OK, then I'll go with linkedid, uniqueid and sequence number.

 Thanks for sharing this on this list


 2013/6/3 Matthew Jordan *mjor...@digium.com* mjor...@digium.com
 On 06/03/2013 11:20 AM, Olivier wrote:
  Hi,
 
  When dealing with CDR SQL tables, I always added an auto-incremented
  cdr_id key as a primary key, just in case provided uniqueid key went
 wrong.
 
  Now I'm facing a situation where I need to insert into a database's
  table and from the dialplan, a reference to the CDR record which is
  currently processed.
 
  So my questions are:
 
  1. Can uniqueid/sequence (or uniqueid/sequence/calldate) bundle be
  safely used as CDR's table primary key  (ie I cannot have any
  uniqueid/sequence combination from one CDR record to match a past
  uniqueid/sequence combination) ?

 Possibly. Things to keep in mind:

 * You can run into uniqueid collisions across multiple systems if you do
 not specify a system name in asterisk.conf or do not specify a unique
 system name in asterisk.conf.
 * You can run into uniqueid collisions if your system clock goes
 backwards for any reason (the uniqueid for a channel happens to use a
 timestamp for its uniqueness)

 Whether or not this is unique enough will be completely dependent on
 your overall system configuration.

 In general, the recommended combination that *should* uniquely specify a
 CDR (when configured correctly) is linkedid (which should be enabled and
 added to your schema), uniqueid, and sequence number, with the asterisk
 system name specified.

 --
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: *http://digium.com* http://digium.com/  *
 http://asterisk.org* http://asterisk.org/



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Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?

2013-06-11 Thread Jonas Kellens

On 06/11/2013 04:39 PM, Richard Mudgett wrote:




On Tue, Jun 11, 2013 at 9:29 AM, Jonas Kellens 
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:


On 06/11/2013 04:12 PM, Matthew J. Roth wrote:

Jonas Kellens wrote:

I notice that it takes 4 to 6 seconds between someone pressing a cipher and
Asterisk continuing inside the dialplan. How come ???

...

Why doesn't Asterisk continue immediately inside the dialplan after having



received the DTMF-input ?


snip

Dialplan :

exten = ivr,1,NoOp()
exten =

ivr,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivr,n,NoOp(${BACKGROUNDSTATUS})
exten = ivr,n,WaitExten(15)
exten = ivr,n,GoTo(restartprompt)

exten = _X,1,Set(choice=${EXTEN})
exten = _X,n,System(echo
'${klantID}','IVR','${choice}','','${CHANNEL:4}','$(date +%s)'
 /var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X,n,other_stuff_I_do

exten = _X.,1,Set(choice=${EXTEN})
exten = _X.,n,System(echo
'${klantID}','IVR','${keuzeID}','','${CHANNEL:4}','$(date +%s)'
 /var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X.,n,other_stuff_I_do


It is waiting for more digits because you have asked it for a possible 
multi-digit exten and it needs to distinguish between the _X and _X. 
patterns.


Richard



Ok thanks.

Any idea how I can resolve this ?

Even if there *can* be more than 1 digit, in case there is only 1 digit 
it should go faster.



Could this dialplan logic be a good solution :

[my-context]
exten = ivr,1,NoOp()
exten = 
ivr,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})

exten = ivr,n,NoOp(${BACKGROUNDSTATUS})
exten = ivr,n,WaitExten(15)
exten = ivr,n,GoTo(restartprompt)

exten = _X,1,Set(choice=${EXTEN})
exten = _X,n,System(echo 
'${klantID}','IVR','${choice}','','${CHANNEL:4}','$(date +%s)'  
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)

exten = _X,n,other_stuff_I_do

exten = ivradvanced,1,NoOp()
exten = 
ivradvanced,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})

exten = ivradvanced,n,NoOp(${BACKGROUNDSTATUS})
exten = ivradvanced,n,WaitExten(15)
exten = ivradvanced,n,GoTo(restartprompt)

exten = _X.,1,Set(choice=${EXTEN})
exten = _X.,n,System(echo 
'${klantID}','IVR','${keuzeID}','','${CHANNEL:4}','$(date +%s)'  
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)

exten = _X.,n,other_stuff_I_do

[another-context]
...
...



Kind regards,

Jonas.


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Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?

2013-06-11 Thread Eric Wieling
The only way to resolve this is to redesign your dialplan so you do not have 
ambiguous matching,   This is not an Asterisk issue, this is an issue with the 
way you designed your dialplan and would apply to any IVR on any system.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, June 11, 2013 10:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Why does it take several seconds to interpret 
DTMF-input ?

On 06/11/2013 04:39 PM, Richard Mudgett wrote:





On Tue, Jun 11, 2013 at 9:29 AM, Jonas Kellens 
jonas.kell...@telenet.be wrote:


On 06/11/2013 04:12 PM, Matthew J. Roth wrote:


Jonas Kellens wrote:

I notice that it takes 4 to 6 seconds between 
someone pressing a cipher and
Asterisk continuing inside the dialplan. How 
come ???

...

Why doesn't Asterisk continue immediately 
inside the dialplan after having

received the DTMF-input ?

snip 



Dialplan :

exten = ivr,1,NoOp()
exten = 
ivr,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivr,n,NoOp(${BACKGROUNDSTATUS})
exten = ivr,n,WaitExten(15)
exten = ivr,n,GoTo(restartprompt)

exten = _X,1,Set(choice=${EXTEN})
exten = _X,n,System(echo 
'${klantID}','IVR','${choice}','','${CHANNEL:4}','$(date +%s)'  
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X,n,other_stuff_I_do

exten = _X.,1,Set(choice=${EXTEN})
exten = _X.,n,System(echo 
'${klantID}','IVR','${keuzeID}','','${CHANNEL:4}','$(date +%s)'  
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X.,n,other_stuff_I_do


It is waiting for more digits because you have asked it for a possible 
multi-digit exten and it needs to distinguish between the _X and _X. patterns.


Richard




Ok thanks.

Any idea how I can resolve this ?

Even if there *can* be more than 1 digit, in case there is only 1 digit it 
should go faster.


Could this dialplan logic be a good solution :

[my-context]
exten = ivr,1,NoOp()
exten = 
ivr,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivr,n,NoOp(${BACKGROUNDSTATUS}) exten = ivr,n,WaitExten(15) exten = 
ivr,n,GoTo(restartprompt)

exten = _X,1,Set(choice=${EXTEN})
exten = _X,n,System(echo 
'${klantID}','IVR','${choice}','','${CHANNEL:4}','$(date +%s)'  
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X,n,other_stuff_I_do

exten = ivradvanced,1,NoOp()
exten = 
ivradvanced,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivradvanced,n,NoOp(${BACKGROUNDSTATUS})
exten = ivradvanced,n,WaitExten(15)
exten = ivradvanced,n,GoTo(restartprompt)

exten = _X.,1,Set(choice=${EXTEN})
exten = _X.,n,System(echo 
'${klantID}','IVR','${keuzeID}','','${CHANNEL:4}','$(date +%s)'  
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X.,n,other_stuff_I_do

[another-context]
...
...



Kind regards,

Jonas.




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Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?

2013-06-11 Thread Patrick Lists

On 06/11/2013 04:44 PM, Jonas Kellens wrote:
[snip]

Ok thanks.

Any idea how I can resolve this ?

Even if there *can* be more than 1 digit, in case there is only 1 digit
it should go faster.


Would it help if they pressed for example 1 followed by the # key?
If not then, as Eric mentioned, redesign your dialplan. Any IVR with a 
double digit amount of options needs some rethinking. IMHO the average 
attention span of a person is such that at option 6 they forgot options 
1 through 5. And if the option explanations last longer than 5 seconds 
it gets even worse.


Regards,
Patrick


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Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?

2013-06-11 Thread Eric Wieling
No.   When you dial 1 the PBX does not know if  it needs to match _X or _X.  

-Original Message-
From: Jonas Kellens [mailto:jonas.kell...@telenet.be] 
Sent: Tuesday, June 11, 2013 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Eric Wieling
Subject: Re: [asterisk-users] Why does it take several seconds to interpret 
DTMF-input ?

On 06/11/2013 04:46 PM, Eric Wieling wrote:


The only way to resolve this is to redesign your dialplan so you do not 
have ambiguous matching,   This is not an Asterisk issue, this is an issue with 
the way you designed your dialplan and would apply to any IVR on any system.


I understand that I need to re-design my dialplan logic.

I gave an example of my re-design in my last post. Would that have been a good 
re-design ?? Or is it still ambiguous ?

I will post it again :



[my-context]
exten = ivr,1,NoOp()
exten = 
ivr,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivr,n,NoOp(${BACKGROUNDSTATUS}) exten = ivr,n,WaitExten(15) exten = 
ivr,n,GoTo(restartprompt)

exten = _X,1,Set(choice=${EXTEN})
exten = _X,n,System(echo 
'${klantID}','IVR','${choice}','','${CHANNEL:4}','$(date +%s)'  
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X,n,other_stuff_I_do

exten = ivradvanced,1,NoOp()
exten = 
ivradvanced,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivradvanced,n,NoOp(${BACKGROUNDSTATUS})
exten = ivradvanced,n,WaitExten(15)
exten = ivradvanced,n,GoTo(restartprompt)

exten = _X.,1,Set(choice=${EXTEN})
exten = _X.,n,System(echo 
'${klantID}','IVR','${keuzeID}','','${CHANNEL:4}','$(date +%s)'  
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X.,n,other_stuff_I_do

[another-context]
...
...


Kind regards,

Jonas.


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Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?

2013-06-11 Thread Jonas Kellens

On 06/11/2013 04:46 PM, Eric Wieling wrote:

The only way to resolve this is to redesign your dialplan so you do not have 
ambiguous matching,   This is not an Asterisk issue, this is an issue with the 
way you designed your dialplan and would apply to any IVR on any system.


I understand that I need to re-design my dialplan logic.

I gave an example of my re-design in my last post. Would that have been 
a good re-design ?? Or is it still ambiguous ?


I will post it again :


[my-context]
exten = ivr,1,NoOp()
exten = 
ivr,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivr,n,NoOp(${BACKGROUNDSTATUS}) exten = ivr,n,WaitExten(15) exten = 
ivr,n,GoTo(restartprompt)

exten = _X,1,Set(choice=${EXTEN})
exten = _X,n,System(echo '${klantID}','IVR','${choice}','','${CHANNEL:4}','$(date 
+%s)' /var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X,n,other_stuff_I_do

exten = ivradvanced,1,NoOp()
exten = 
ivradvanced,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivradvanced,n,NoOp(${BACKGROUNDSTATUS})
exten = ivradvanced,n,WaitExten(15)
exten = ivradvanced,n,GoTo(restartprompt)

exten = _X.,1,Set(choice=${EXTEN})
exten = _X.,n,System(echo '${klantID}','IVR','${keuzeID}','','${CHANNEL:4}','$(date 
+%s)' /var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X.,n,other_stuff_I_do

[another-context]
...
...



Kind regards,

Jonas.
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Re: [asterisk-users] Where is HAVE_NEWLOCALE set?

2013-06-11 Thread Tzafrir Cohen
On Mon, Jun 10, 2013 at 04:06:27PM -0400, D'Arcy J.M. Cain wrote:
 I am trying to build Asterisk on a NetBSD system but I am running into
 two problems.  The first only happens on an installation built from
 NetBSD HEAD.  The config variable HAVE_NEWLOCALE is erroneously set
 during configure but this system does not have newlocale().  I can't
 seem to find where this gets set to true.

What version of asterisk is it? It is set by the configure script (look
for 'newlocale' in configure.ac).

 
 Interestingly a stable release of NetBSD does not have this issue
 although it still has the second issue which I will start a separate
 thread for.

-- 
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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

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Re: [asterisk-users] DTLSv1_method on NetBSD

2013-06-11 Thread Tzafrir Cohen
On Mon, Jun 10, 2013 at 04:10:23PM -0400, D'Arcy J.M. Cain wrote:
 This is the second issue I found while trying to install Asterisk on a
 NetBSD box.  I can't load the rtp module because HAVE_OPENSSL_SRTP
 seems to be set.  Is there some way to simply force this variab;e to be
 unset from a configuration variable?

Do you have OpenSSL?

Again, what version of Asterisk? What version of NetBSD? People may read
this in the future, so please provide a version number rather than
stable.

-- 
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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

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Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?

2013-06-11 Thread Matthew J. Roth
Jonas Kellens wrote: 
 
 Even if there *can* be more than 1 digit, in case there is only 1 digit it
 should go faster.


Jonas,

Use the TIMEOUT function to set the maximum amount of time permitted between
digits when the user is typing in DTMF.  As you've discovered, the default is 5
seconds.

The following example reduces it to 2 seconds:

exten = ivr,1,NoOp()
exten = ivr,n,Set(TIMEOUT(digit)=2)
exten = 
ivr,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivr,n,NoOp(${BACKGROUNDSTATUS})
exten = ivr,n,WaitExten(15)
exten = ivr,n,GoTo(restartprompt)


Enter 'core show function TIMEOUT' at the Asterisk CLI for more information 
about
the TIMEOUT function.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] Where is HAVE_NEWLOCALE set?

2013-06-11 Thread D'Arcy J.M. Cain
On Tue, 11 Jun 2013 18:42:07 +0300
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
 On Mon, Jun 10, 2013 at 04:06:27PM -0400, D'Arcy J.M. Cain wrote:
  I am trying to build Asterisk on a NetBSD system but I am running
  into two problems.  The first only happens on an installation built
  from NetBSD HEAD.  The config variable HAVE_NEWLOCALE is
  erroneously set during configure but this system does not have
  newlocale().  I can't seem to find where this gets set to true.
 
 What version of asterisk is it? It is set by the configure script
 (look for 'newlocale' in configure.ac).

It's version 11.4.0 which is supposedly the latest.  Yes, it is in
AC_CHECK_FUNCS.  I just don't understand why it thinks that it exists.

  Interestingly a stable release of NetBSD does not have this issue

By stable I mean NetBSD 6.0.1 (GENERIC) amd64.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
Voip: sip:da...@vex.net

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Re: [asterisk-users] announcement to be played for attended transfer call

2013-06-11 Thread jg
So, B transfers the call and after bridging to C, B should get an 
announcement.


This is just an idea:
See whether you can dispatch the termination of the call leg B-C by 
evaluating the DIALSTATUS variable. I am not sure whether you can see 
this inside the dialplan, but you should get the event via AMI. This is 
only the 1st part of the solution.


A general solution would require a lot of things or may not be possible 
at all as you can transfer calls not only via Asterisk using DTMF 
signalling, but also the SIP phones themselves might be capable of 
transferring calls, thereby circumventing Asterisk.


jg



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Re: [asterisk-users] DTLSv1_method on NetBSD

2013-06-11 Thread D'Arcy J.M. Cain
On Tue, 11 Jun 2013 18:43:55 +0300
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
 On Mon, Jun 10, 2013 at 04:10:23PM -0400, D'Arcy J.M. Cain wrote:
  This is the second issue I found while trying to install Asterisk
  on a NetBSD box.  I can't load the rtp module because
  HAVE_OPENSSL_SRTP seems to be set.  Is there some way to simply
  force this variab;e to be unset from a configuration variable?
 
 Do you have OpenSSL?

Yes, it is in the base system.  OpenSSL 1.0.1c 10 May 2012.

 Again, what version of Asterisk? What version of NetBSD? People may

Asterisk 11.4.0.  NetBSD 6.0.1 (GENERIC) amd64

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Voip: sip:da...@vex.net

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Re: [asterisk-users] announcement to be played for attended transfer call

2013-06-11 Thread jg

Since Dial() might not return, DIALSTATUS cannot be used.

I checked the various AMI events and you'll see a bunch of Newchannel, 
Hangup, Bridge, Unlink, and Masquerade events when transferring calls. 
You could use this to originate a call with the announcement for B. This 
is ugly, but if B's phone has intercom capabilities, this might be a 
quick and dirty solution that works.


On the other hand, if B does the transfer and knows about the state of 
the transfer, why should there be an extra announcement?


jg

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[asterisk-users] CDR_MYSQL

2013-06-11 Thread Nicholas Hart
I need to install cdr_mysql.so module for logging call to mysql.  I have
the source file cdr_mysql.c only.  Can someone explain the steps needed to
get this module compiled and working in Asterisk 1.8.22.0 on CentOS.

Thanks.
Nick
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Re: [asterisk-users] CDR_MYSQL

2013-06-11 Thread jg
How about 
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DB.html 
?


jg

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Re: [asterisk-users] + dialplan

2013-06-11 Thread Jonson Player
Hello Adam,

Thank you very much for your info.

Regards,
Jonson.

On Tue, Jun 11, 2013 at 12:34 AM, ad...@3a.hu wrote:

 Hi,


 On 06/10/2013 22:26, Jonson Player wrote:

 Some users of main use + instead of 00 for international dial. Is there
 any solution for this problem?


 swap the + sign to double zeros if your provider can't handle it

 ; normal 00 prefix
 exten = _00ZZXXX.,1,Macro(**beforealldials)
 exten = _00ZZXXX.,n,Dial(SIP/${**EXTEN}@${OUTGOING_LINE})
 exten = _00ZZXXX.,n,Hangup()

 ; swap + prefix to 00
 exten = _+ZZXXX.,1,Macro(**beforealldials)
 exten = _+ZZXXX.,n,Dial(SIP/00${**EXTEN:1}@${OUTGOING_LINE})
 exten = _+ZZXXX.,n,Hangup()

 regards
 adam



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Re: [asterisk-users] A problem with IAX2

2013-06-11 Thread Mordechay Kaganer
B.H.

On Jun 11, 2013 5:15 PM, Steve Totaro stot...@totarotechnologies.com
wrote:




 On Tue, Jun 11, 2013 at 8:32 AM, Mordechay Kaganer mkaga...@gmail.com
wrote:

 B.H.

 Hello!

 We have several Asterik boxes that are connected to PSTN using PRI cards
and they are interconnected using IAX2 trunks so that incoming calls are
delivered from PSTN to the servers they belong to.

 In past we were using asterisk 1.4 on the server that is receiving IAX
connections and everything worked as expected. Recently, we have switched
to a newer box with asterisk 1.8.22 and then we began to experience
sometimes a strange problem:

 At some point of time, incoming IAX connections begin to get refused by
the server and we get the following messages in the logs:

 WARNING[] chan_iax2.c: Too much delay in IAX2 calltoken timestamp
from address X.X.X.X

 where X.X.X.X is the IP of the PSTN-IAX gateways and all the incoming
calls start to be rejected.

 Direct PSTN calls (both incoming and outgoing) to the same server work
OK. The only solution that helps is to kill the asterisk and restart it.

 All the servers are connected to the same LAN segment, with gigabit
switch, there is no problems with the network. No packet loss.

 There's already bug report present with very similar issue, but it is
suspended and, like stated there, the problem is very hard to reproduce.

 See: https://issues.asterisk.org/jira/browse/ASTERISK-21762


 --
 משיח NOW!


 Use SIP and never look back.

 Thanks,
 Steve Totaro

 --


Thanks, that's what i actually going to do.

But does this mean that IAX is obsolete? Actually i have selected IAX in
the first place because it looks like more native for asterisk, so i
thought it would be more suitable as a protocol to interconnect asterisk
boxes...
_
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Re: [asterisk-users] A problem with IAX2

2013-06-11 Thread Fabio Moretti

  
  
hi,

I've solved various iax2 problem mentioning calltoken when I put
these lines in the iax configuration:

requirecalltoken=no
calltokenoptional=0.0.0.0/0.0.0.0

bye

Il 11/06/2013 19:25, Mordechay Kaganer
  scrisse:


  B.H.
  On Jun 11, 2013 5:15 PM, "Steve Totaro" stot...@totarotechnologies.com
wrote:




 On Tue, Jun 11, 2013 at 8:32 AM, Mordechay Kaganer mkaga...@gmail.com
wrote:

 B.H.

 Hello!

 We have several Asterik boxes that are connected to
PSTN using PRI cards and they are interconnected using IAX2
trunks so that incoming calls are delivered from PSTN to the
servers they belong to.

 In past we were using asterisk 1.4 on the server that
is receiving IAX connections and everything worked as expected.
Recently, we have switched to a newer box with asterisk 1.8.22
and then we began to experience sometimes a strange problem:

 At some point of time, incoming IAX connections begin
to get refused by the server and we get the following messages
in the logs:

 WARNING[] chan_iax2.c: Too much delay in IAX2
calltoken timestamp from address X.X.X.X

 where X.X.X.X is the IP of the PSTN-IAX gateways
and all the incoming calls start to be rejected.

 Direct PSTN calls (both incoming and outgoing) to the
same server work OK. The only solution that helps is to kill the
asterisk and restart it.

 All the servers are connected to the same LAN segment,
with gigabit switch, there is no problems with the network. No
packet loss.

 There's already bug report present with very similar
issue, but it is "suspended" and, like stated there, the problem
is very hard to reproduce.

 See:https://issues.asterisk.org/jira/browse/ASTERISK-21762


 -- 
  NOW!


 Use SIP and never look back.

 Thanks,
 Steve Totaro

 --

  Thanks, that's what i actually going to do.
  But does this mean that IAX is obsolete? Actually i have
selected IAX in the first place because it looks like more
"native" for asterisk, so i thought it would be more suitable as
a protocol to interconnect asterisk boxes...
_
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-- 
  

  
  
  Fabio
Moretti  
  Gerente de Sistemas 
   www.tecytal.com  
   0800 8780 
(+598) 248 77921  
  

  

  


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