[asterisk-users] pipe character in CDR user field
I'm trying to set a CDR userfield to a custom value. This value may contain a '|' but it's really just part of the value. However, Asterisk keeps warning me about the application delimiter not being a pipe. It's NOT an application delimiter (it's just part of a variable value) so I'm expecting Asterisk not to warn me about it. Is it expected behavior? Why? See the following log: SIP/4053-007bAGI Rx EXEC Set CDR(userfield)=|usr_r=vieri -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=|usr_r=vieri) [Nov 29 10:53:08] WARNING[4815]: pbx.c:1563 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Set(CDR(userfield)=|usr_r=vieri)) SIP/4053-007bAGI Tx 200 result=0 SIP/4053-007dAGI Rx EXEC Set CDR(userfield)=\|usr_r=vieri\ -- AGI Script Executing Application: (Set) Options: (CDR(userfield)=|usr_r=vieri) [Nov 29 10:54:57] WARNING[4838]: pbx.c:1563 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Set(CDR(userfield)=|usr_r=vieri)) SIP/4053-007dAGI Tx 200 result=0 Thanks, Vieri -- _ -- 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] Disappearing Call Files / Two threads dealing with my call files
Hello,
I noticed that when i move a call file to outgoing directory, two asterisk
threads are dealing with it.
]# grep FAX_44731.call /var/log/asterisk/full.2
[Nov 27 09:23:10] WARNING[26842] pbx_spool.c: Unable to set utime on
/var/spool/asterisk/outgoing/FAX_44731.call: Operation not permitted
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:1]
System(DAHDI/i1/0312xxx-b08, echo Set: UNIQUEID=1354000990.39861
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:3]
System(DAHDI/i1/0312xxx-b08, echo Set: FAXSTATUS=SUCCESS
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: At least one of app or
extension must be specified, along with tech and dest in file
/var/spool/asterisk/outgoing/FAX_44731.call
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: Invalid file contents in
/var/spool/asterisk/outgoing/FAX_44731.call, deleting
As you see there are two thread dealing with my call file. Now let's
inspect the thread 18852.
]# grep \[18852\] /var/log/asterisk/full.2
[Nov 27 09:23:10] VERBOSE[18852] pbx_spool.c: -- Attempting call on
DAHDI/g0/0312xxx for s@asteriskgw_fax:1 (Retry 1)
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: sig_pri_request 5
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: CALLER NAME: NUM: 90312xxx
[Nov 27 09:23:10] VERBOSE[18852] sig_pri.c: -- Requested transfer
capability: 0x00 - SPEECH
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:1]
System(DAHDI/i1/0312xxx-b08, echo Set: UNIQUEID=1354000990.39861
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:2]
SendFAX(DAHDI/i1/0312xxx-b08, /tmp/Qg90Ox5YGF5kYkJu.tif,zdfs) in
new stack
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: -- Channel
'DAHDI/i1/0312xxx-b08' sending FAX:
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: --
/tmp/Qg90Ox5YGF5kYkJu.tif
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:3]
System(DAHDI/i1/0312xxx-b08, echo Set: FAXSTATUS=SUCCESS
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Auto fallthrough, channel
'DAHDI/i1/0312xxx-b08' status is 'UNKNOWN'
[Nov 27 09:25:33] VERBOSE[18852] chan_dahdi.c: -- Hungup
'DAHDI/i1/0312xxx-b08'
[Nov 27 09:25:33] NOTICE[18852] pbx_spool.c: Call completed to
DAHDI/g0/0312xxx
It seems that the thread 18852 executes it normally but the thread 26842
deletes my call file. And when I inspected the asterisk log file, i saw
that the thread 26842 is deleting all my call files.
Here is my custom_extensions.conf file:
[asteriskgw_fax]
exten = s,1,System(echo Set: UNIQUEID=${CDR(uniqueid)}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
exten = s,2,SendFAX(${FAXFILE},zdfs)
exten = s,3,System(echo Set: FAXSTATUS=${FAXSTATUS}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
And here is a sample of call file:
Channel: DAHDI/g0/0312xxx
MaxRetries: 0
RetryTime: 60
Context: asteriskgw_fax
Extension: s
Set: FAXFILE=/tmp/8Mg3yihXahZVejDf.tif
Set: ID=44884
Callerid: 90312xxx
Archive: Yes
--
Necati DEMİR
--
_
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Re: [asterisk-users] Disappearing Call Files / Two threads dealing with my call files
There's no priority in your call file.
Sent from my iPhone
On 29/11/2012, at 11:12 PM, Necati Demir nde...@demir.web.tr wrote:
Hello,
I noticed that when i move a call file to outgoing directory, two asterisk
threads are dealing with it.
]# grep FAX_44731.call /var/log/asterisk/full.2
[Nov 27 09:23:10] WARNING[26842] pbx_spool.c: Unable to set utime on
/var/spool/asterisk/outgoing/FAX_44731.call: Operation not permitted
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:1]
System(DAHDI/i1/0312xxx-b08, echo Set: UNIQUEID=1354000990.39861
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:3]
System(DAHDI/i1/0312xxx-b08, echo Set: FAXSTATUS=SUCCESS
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: At least one of app or
extension must be specified, along with tech and dest in file
/var/spool/asterisk/outgoing/FAX_44731.call
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: Invalid file contents in
/var/spool/asterisk/outgoing/FAX_44731.call, deleting
As you see there are two thread dealing with my call file. Now let's inspect
the thread 18852.
]# grep \[18852\] /var/log/asterisk/full.2
[Nov 27 09:23:10] VERBOSE[18852] pbx_spool.c: -- Attempting call on
DAHDI/g0/0312xxx for s@asteriskgw_fax:1 (Retry 1)
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: sig_pri_request 5
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: CALLER NAME: NUM: 90312xxx
[Nov 27 09:23:10] VERBOSE[18852] sig_pri.c: -- Requested transfer
capability: 0x00 - SPEECH
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:1]
System(DAHDI/i1/0312xxx-b08, echo Set: UNIQUEID=1354000990.39861
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:2]
SendFAX(DAHDI/i1/0312xxx-b08, /tmp/Qg90Ox5YGF5kYkJu.tif,zdfs) in new
stack
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: -- Channel
'DAHDI/i1/0312xxx-b08' sending FAX:
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: --
/tmp/Qg90Ox5YGF5kYkJu.tif
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:3]
System(DAHDI/i1/0312xxx-b08, echo Set: FAXSTATUS=SUCCESS
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Auto fallthrough, channel
'DAHDI/i1/0312xxx-b08' status is 'UNKNOWN'
[Nov 27 09:25:33] VERBOSE[18852] chan_dahdi.c: -- Hungup
'DAHDI/i1/0312xxx-b08'
[Nov 27 09:25:33] NOTICE[18852] pbx_spool.c: Call completed to
DAHDI/g0/0312xxx
It seems that the thread 18852 executes it normally but the thread 26842
deletes my call file. And when I inspected the asterisk log file, i saw that
the thread 26842 is deleting all my call files.
Here is my custom_extensions.conf file:
[asteriskgw_fax]
exten = s,1,System(echo Set: UNIQUEID=${CDR(uniqueid)}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
exten = s,2,SendFAX(${FAXFILE},zdfs)
exten = s,3,System(echo Set: FAXSTATUS=${FAXSTATUS}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
And here is a sample of call file:
Channel: DAHDI/g0/0312xxx
MaxRetries: 0
RetryTime: 60
Context: asteriskgw_fax
Extension: s
Set: FAXFILE=/tmp/8Mg3yihXahZVejDf.tif
Set: ID=44884
Callerid: 90312xxx
Archive: Yes
--
Necati DEMİR
--
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Re: [asterisk-users] Disappearing Call Files / Two threads dealing with my call files
Should I use priority in call files? How the lack of priority causes this
problem?
On 29 November 2012 12:48, Matt Riddell (lists) li...@venturevoip.comwrote:
There's no priority in your call file.
Sent from my iPhone
On 29/11/2012, at 11:12 PM, Necati Demir nde...@demir.web.tr wrote:
Hello,
I noticed that when i move a call file to outgoing directory, two
asterisk threads are dealing with it.
]# grep FAX_44731.call /var/log/asterisk/full.2
[Nov 27 09:23:10] WARNING[26842] pbx_spool.c: Unable to set utime on
/var/spool/asterisk/outgoing/FAX_44731.call: Operation not permitted
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:1] System(DAHDI/i1/0312xxx-b08, echo Set:
UNIQUEID=1354000990.39861 /var/spool/asterisk/outgoing/FAX_44731.call)
in new stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:3] System(DAHDI/i1/0312xxx-b08, echo Set:
FAXSTATUS=SUCCESS /var/spool/asterisk/outgoing/FAX_44731.call) in new
stack
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: At least one of app or
extension must be specified, along with tech and dest in file
/var/spool/asterisk/outgoing/FAX_44731.call
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: Invalid file contents in
/var/spool/asterisk/outgoing/FAX_44731.call, deleting
As you see there are two thread dealing with my call file. Now let's
inspect the thread 18852.
]# grep \[18852\] /var/log/asterisk/full.2
[Nov 27 09:23:10] VERBOSE[18852] pbx_spool.c: -- Attempting call on
DAHDI/g0/0312xxx for s@asteriskgw_fax:1 (Retry 1)
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: sig_pri_request 5
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: CALLER NAME: NUM: 90312xxx
[Nov 27 09:23:10] VERBOSE[18852] sig_pri.c: -- Requested transfer
capability: 0x00 - SPEECH
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:1] System(DAHDI/i1/0312xxx-b08, echo Set:
UNIQUEID=1354000990.39861 /var/spool/asterisk/outgoing/FAX_44731.call)
in new stack
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:2] SendFAX(DAHDI/i1/0312xxx-b08,
/tmp/Qg90Ox5YGF5kYkJu.tif,zdfs) in new stack
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: -- Channel
'DAHDI/i1/0312xxx-b08' sending FAX:
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: --
/tmp/Qg90Ox5YGF5kYkJu.tif
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:3] System(DAHDI/i1/0312xxx-b08, echo Set:
FAXSTATUS=SUCCESS /var/spool/asterisk/outgoing/FAX_44731.call) in new
stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Auto fallthrough, channel
'DAHDI/i1/0312xxx-b08' status is 'UNKNOWN'
[Nov 27 09:25:33] VERBOSE[18852] chan_dahdi.c: -- Hungup
'DAHDI/i1/0312xxx-b08'
[Nov 27 09:25:33] NOTICE[18852] pbx_spool.c: Call completed to
DAHDI/g0/0312xxx
It seems that the thread 18852 executes it normally but the thread 26842
deletes my call file. And when I inspected the asterisk log file, i saw
that the thread 26842 is deleting all my call files.
Here is my custom_extensions.conf file:
[asteriskgw_fax]
exten = s,1,System(echo Set: UNIQUEID=${CDR(uniqueid)}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
exten = s,2,SendFAX(${FAXFILE},zdfs)
exten = s,3,System(echo Set: FAXSTATUS=${FAXSTATUS}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
And here is a sample of call file:
Channel: DAHDI/g0/0312xxx
MaxRetries: 0
RetryTime: 60
Context: asteriskgw_fax
Extension: s
Set: FAXFILE=/tmp/8Mg3yihXahZVejDf.tif
Set: ID=44884
Callerid: 90312xxx
Archive: Yes
--
Necati DEMİR
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Necati DEMİR
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Re: [asterisk-users] Disappearing Call Files / Two threads dealing with my call files
Priority is a required parameter. In your call file you are telling Asterisk
to
Channel: DAHDI/g0/0312xxx
MaxRetries: 0
RetryTime: 60
Context: asteriskgw_fax
Extension: s
Go to context asteriskgw_fax, extension s. Priority tells Asterisk where to
start in asteriskgw_fax. Since C would assume 0 and contexts start with 1,
priority: 1 tells it to go to line 1. Another use for this would be to tell
Asterisk to start further down to skip a wait or something.
Sample:
[asteriskgw_fax]
Exten = s,1,answer()
Exten = s,n,wait(5)
Exten = s,n,playback(sending-fax)
You could use priority 1 for DAHDI to compensate for PSTN delays and priority 3
for SIP calls.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Necati Demir
Sent: Thursday, November 29, 2012 8:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Disappearing Call Files / Two threads dealing
with my call files
Should I use priority in call files? How the lack of priority causes this
problem?
On 29 November 2012 12:48, Matt Riddell (lists) li...@venturevoip.com wrote:
There's no priority in your call file.
Sent from my iPhone
On 29/11/2012, at 11:12 PM, Necati Demir nde...@demir.web.tr wrote:
Hello,
I noticed that when i move a call file to outgoing directory, two asterisk
threads are dealing with it.
]# grep FAX_44731.call /var/log/asterisk/full.2
[Nov 27 09:23:10] WARNING[26842] pbx_spool.c: Unable to set utime on
/var/spool/asterisk/outgoing/FAX_44731.call: Operation not permitted
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:1]
System(DAHDI/i1/0312xxx-b08, echo Set: UNIQUEID=1354000990.39861
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:3]
System(DAHDI/i1/0312xxx-b08, echo Set: FAXSTATUS=SUCCESS
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: At least one of app or
extension must be specified, along with tech and dest in file
/var/spool/asterisk/outgoing/FAX_44731.call
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: Invalid file contents in
/var/spool/asterisk/outgoing/FAX_44731.call, deleting
As you see there are two thread dealing with my call file. Now let's inspect
the thread 18852.
]# grep \[18852\] /var/log/asterisk/full.2
[Nov 27 09:23:10] VERBOSE[18852] pbx_spool.c: -- Attempting call on
DAHDI/g0/0312xxx for s@asteriskgw_fax:1 (Retry 1)
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: sig_pri_request 5
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: CALLER NAME: NUM: 90312xxx
[Nov 27 09:23:10] VERBOSE[18852] sig_pri.c: -- Requested transfer
capability: 0x00 - SPEECH
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:1]
System(DAHDI/i1/0312xxx-b08, echo Set: UNIQUEID=1354000990.39861
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:2]
SendFAX(DAHDI/i1/0312xxx-b08, /tmp/Qg90Ox5YGF5kYkJu.tif,zdfs) in new
stack
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: -- Channel
'DAHDI/i1/0312xxx-b08' sending FAX:
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: --
/tmp/Qg90Ox5YGF5kYkJu.tif
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing [s@asteriskgw_fax:3]
System(DAHDI/i1/0312xxx-b08, echo Set: FAXSTATUS=SUCCESS
/var/spool/asterisk/outgoing/FAX_44731.call) in new stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Auto fallthrough, channel
'DAHDI/i1/0312xxx-b08' status is 'UNKNOWN'
[Nov 27 09:25:33] VERBOSE[18852] chan_dahdi.c: -- Hungup
'DAHDI/i1/0312xxx-b08'
[Nov 27 09:25:33] NOTICE[18852] pbx_spool.c: Call completed to
DAHDI/g0/0312xxx
It seems that the thread 18852 executes it normally but the thread 26842
deletes my call file. And when I inspected the asterisk log file, i saw that
the thread 26842 is deleting all my call files.
Here is my custom_extensions.conf file:
[asteriskgw_fax]
exten = s,1,System(echo Set: UNIQUEID=${CDR(uniqueid)}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
exten = s,2,SendFAX(${FAXFILE},zdfs)
exten = s,3,System(echo Set: FAXSTATUS=${FAXSTATUS}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
And here is a sample of call file:
Channel: DAHDI/g0/0312xxx
MaxRetries: 0
RetryTime: 60
Context: asteriskgw_fax
Extension: s
Set: FAXFILE=/tmp/8Mg3yihXahZVejDf.tif
Set: ID=44884
Callerid: 90312xxx
Archive: Yes
--
Necati DEMİR
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
To
Re: [asterisk-users] Disappearing Call Files / Two threads dealing with my call files
Thanks, i will add priority and see the results.
On 29 November 2012 17:00, Danny Nicholas da...@debsinc.com wrote:
Priority is a required parameter. In your call file you are telling
Asterisk to
Channel: DAHDI/g0/0312xxx
MaxRetries: 0
RetryTime: 60
Context: asteriskgw_fax
Extension: s
Go to context asteriskgw_fax, extension s. Priority tells Asterisk where
to start in asteriskgw_fax. Since C would assume 0 and contexts start with
1, priority: 1 tells it to go to line 1. Another use for this would be to
tell Asterisk to start further down to skip a wait or something.
Sample:
[asteriskgw_fax]
Exten = s,1,answer()
Exten = s,n,wait(5)
Exten = s,n,playback(sending-fax)
** **
You could use priority 1 for DAHDI to compensate for PSTN delays and
priority 3 for SIP calls.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Necati Demir
*Sent:* Thursday, November 29, 2012 8:50 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Disappearing Call Files / Two threads
dealing with my call files
** **
** **
Should I use priority in call files? How the lack of priority causes this
problem?
** **
On 29 November 2012 12:48, Matt Riddell (lists) li...@venturevoip.com
wrote:
There's no priority in your call file.
Sent from my iPhone
On 29/11/2012, at 11:12 PM, Necati Demir nde...@demir.web.tr wrote:
Hello,
I noticed that when i move a call file to outgoing directory, two
asterisk threads are dealing with it.
]# grep FAX_44731.call /var/log/asterisk/full.2
[Nov 27 09:23:10] WARNING[26842] pbx_spool.c: Unable to set utime on
/var/spool/asterisk/outgoing/FAX_44731.call: Operation not permitted
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:1] System(DAHDI/i1/0312xxx-b08, echo Set:
UNIQUEID=1354000990.39861 /var/spool/asterisk/outgoing/FAX_44731.call)
in new stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:3] System(DAHDI/i1/0312xxx-b08, echo Set:
FAXSTATUS=SUCCESS /var/spool/asterisk/outgoing/FAX_44731.call) in new
stack
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: At least one of app or
extension must be specified, along with tech and dest in file
/var/spool/asterisk/outgoing/FAX_44731.call
[Nov 27 09:25:33] WARNING[26842] pbx_spool.c: Invalid file contents in
/var/spool/asterisk/outgoing/FAX_44731.call, deleting
As you see there are two thread dealing with my call file. Now let's
inspect the thread 18852.
]# grep \[18852\] /var/log/asterisk/full.2
[Nov 27 09:23:10] VERBOSE[18852] pbx_spool.c: -- Attempting call on
DAHDI/g0/0312xxx for s@asteriskgw_fax:1 (Retry 1)
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: sig_pri_request 5
[Nov 27 09:23:10] DEBUG[18852] sig_pri.c: CALLER NAME: NUM: 90312xxx
[Nov 27 09:23:10] VERBOSE[18852] sig_pri.c: -- Requested transfer
capability: 0x00 - SPEECH
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:1] System(DAHDI/i1/0312xxx-b08, echo Set:
UNIQUEID=1354000990.39861 /var/spool/asterisk/outgoing/FAX_44731.call)
in new stack
[Nov 27 09:23:25] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:2] SendFAX(DAHDI/i1/0312xxx-b08,
/tmp/Qg90Ox5YGF5kYkJu.tif,zdfs) in new stack
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: -- Channel
'DAHDI/i1/0312xxx-b08' sending FAX:
[Nov 27 09:23:25] VERBOSE[18852] res_fax.c: --
/tmp/Qg90Ox5YGF5kYkJu.tif
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Executing
[s@asteriskgw_fax:3] System(DAHDI/i1/0312xxx-b08, echo Set:
FAXSTATUS=SUCCESS /var/spool/asterisk/outgoing/FAX_44731.call) in new
stack
[Nov 27 09:25:33] VERBOSE[18852] pbx.c: -- Auto fallthrough, channel
'DAHDI/i1/0312xxx-b08' status is 'UNKNOWN'
[Nov 27 09:25:33] VERBOSE[18852] chan_dahdi.c: -- Hungup
'DAHDI/i1/0312xxx-b08'
[Nov 27 09:25:33] NOTICE[18852] pbx_spool.c: Call completed to
DAHDI/g0/0312xxx
It seems that the thread 18852 executes it normally but the thread 26842
deletes my call file. And when I inspected the asterisk log file, i saw
that the thread 26842 is deleting all my call files.
Here is my custom_extensions.conf file:
[asteriskgw_fax]
exten = s,1,System(echo Set: UNIQUEID=${CDR(uniqueid)}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
exten = s,2,SendFAX(${FAXFILE},zdfs)
exten = s,3,System(echo Set: FAXSTATUS=${FAXSTATUS}
/var/spool/asterisk/outgoing/FAX_${ID}.call)
And here is a sample of call file:
Channel: DAHDI/g0/0312xxx
MaxRetries: 0
RetryTime: 60
Context: asteriskgw_fax
Extension: s
Set: FAXFILE=/tmp/8Mg3yihXahZVejDf.tif
Set: ID=44884
Callerid: 90312xxx
Archive: Yes
--
Necati DEMİR
--
Re: [asterisk-users] pipe character in CDR user field
On Nov 29, 2012, at 3:54 AM, Vieri wrote:
I'm trying to set a CDR userfield to a custom value. This value may contain a
'|' but it's really just part of the value.
However, Asterisk keeps warning me about the application delimiter not being
a pipe.
It's NOT an application delimiter (it's just part of a variable value) so I'm
expecting Asterisk not to warn me about it.
Is it expected behavior?
Well, you certainly didn't expect it. I would not have expected it either :-D
Why?
So, back in the early days, Asterisk inconsistently used different delimiters
depending upon context. Sometimes it used pipes, sometimes commas.
Inconsistency is never good, so we picked a winner (commas) and put in code to
look for the loser (pipes) in the dial plan and warn if they show up. ('We' in
this case would be Asterisk developers of long ago, by the way).
The code that does this checking[1] isn't the smartest code in the world. It
basically looks at the data passed to the application, and if it contains a
pipe, and no comma, and warnings are enabled, it warns you.
So you could disable warnings, but that would turn off other warnings that
might be useful.
Another option would be a small hack in your dial plan: add a comma.
MSet(CDR(userfield)=|usr_r=vieri,PIPE_HACK=true) ; Asterisk warns if it
sees a pipe without a comma.
Since you're not trying to use a pipe as a delimiter, displaying the warning is
a mistake in Asterisk. The whole pipe/comma thing happened so long ago[2], it's
time to just lose the warning altogether.
[1]:
https://code.asterisk.org/code/browse/asterisk/branches/11/main/pbx.c?u=3r=376690#to1583
[2]: https://code.asterisk.org/code/changelog/asterisk?cs=188210
See the following log:
SIP/4053-007bAGI Rx EXEC Set CDR(userfield)=|usr_r=vieri
-- AGI Script Executing Application: (Set) Options:
(CDR(userfield)=|usr_r=vieri)
[Nov 29 10:53:08] WARNING[4815]: pbx.c:1563 pbx_exec: The application
delimiter is now the comma, not the pipe. Did you forget to convert your
dialplan? (Set(CDR(userfield)=|usr_r=vieri))
SIP/4053-007bAGI Tx 200 result=0
SIP/4053-007dAGI Rx EXEC Set CDR(userfield)=\|usr_r=vieri\
-- AGI Script Executing Application: (Set) Options:
(CDR(userfield)=|usr_r=vieri)
[Nov 29 10:54:57] WARNING[4838]: pbx.c:1563 pbx_exec: The application
delimiter is now the comma, not the pipe. Did you forget to convert your
dialplan? (Set(CDR(userfield)=|usr_r=vieri))
SIP/4053-007dAGI Tx 200 result=0
Thanks,
Vieri
--
David M. Lee
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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[asterisk-users] Questions about extension.conf
Hello
I have been reading the sample extension.conf
;###
[outbound-freenum2]
; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
;
exten = _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same = n,Set(SUFFIX=${CUT(EXTEN,*,2-)}); make
sure the suffix is all digits as well
same = n,GotoIf($[${FILTER(0-9,${SUFFIX})} != ${SUFFIX}]?fn-CONGESTION,1)
;
filter out bad characters per the README-SERIOUSLY.best-practices.txt document
same = n,Set(TIMEOUT(absolute)=10800)
same = n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ;
perform our lookup with freenum.org
same = n,GotoIf($[${isnresult} != ]?from)
same = n,Set(DIALSTATUS=CONGESTION)
same = n,Goto(fn-CONGESTION,1)
same = n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} = ]?dial) ; check
if we set the FREENUMDOMAIN global variable in [global]
same = n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ;if
we did set it, then we'll use it for our outbound dialing domain
same = n(dial),Dial(SIP/${isnresult},40)
same = n,Goto(fn-${DIALSTATUS},1)
exten = fn-BUSY,1,Busy()
exten = _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same = n,Congestion()
;##
According to
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf;
Syntax for defining a context: keywords exten, include, ignorepat and switch.
same is not mentioned in this wiki.
There is a part of dial plan from sample extension.conf above. My Question is
how same = key word works .
Thanks
--
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Re: [asterisk-users] Questions about extension.conf
As I understand it, same = is a way to shorthand your list of the other
keywords. In the example you posted, you save 4 keystrokes for each line you
enter; not a lot of savings for this short example, but put it in a 1000+
line dialplan and it's quite a time-saver.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shitian Long
Sent: Thursday, November 29, 2012 10:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Questions about extension.conf
Hello
I have been reading the sample extension.conf
;###
[outbound-freenum2]
; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
;
exten = _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same = n,Set(SUFFIX=${CUT(EXTEN,*,2-)});
make sure the suffix is all digits as well
same = n,GotoIf($[${FILTER(0-9,${SUFFIX})} !=
${SUFFIX}]?fn-CONGESTION,1)
;
filter out bad characters per the README-SERIOUSLY.best-practices.txt
document
same = n,Set(TIMEOUT(absolute)=10800)
same = n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ;
perform our lookup with freenum.org
same = n,GotoIf($[${isnresult} != ]?from)
same = n,Set(DIALSTATUS=CONGESTION)
same = n,Goto(fn-CONGESTION,1)
same = n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} = ]?dial) ;
check if we set the FREENUMDOMAIN global variable in [global]
same = n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ;
if we did set it, then we'll use it for our outbound dialing domain
same = n(dial),Dial(SIP/${isnresult},40)
same = n,Goto(fn-${DIALSTATUS},1)
exten = fn-BUSY,1,Busy()
exten = _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same = n,Congestion()
;##
According to
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf;
Syntax for defining a context: keywords exten, include, ignorepat and
switch. same is not mentioned in this wiki.
There is a part of dial plan from sample extension.conf above. My Question
is how same = key word works .
Thanks
--
_
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Re: [asterisk-users] Questions about extension.conf
Shitian Long wrote 29.11.2012 18:40: There is a part of dial plan from sample extension.conf above. My Question is how same = key word works . Thanks same is used for complex templates, if you don't want to copy previous line or afraid you can make a typo. exten = _1XXNXXX,1,Answer same = n,HangUp is the substitution for: exten = _1XXNXXX,1,Answer exten = _1XXNXXX,n,HangUp Also, it makes grepping the particular exten in a file a lot easier, and if you want to change some template for exten which has 50 lines, you don't have to edit all 50 of them. -- With Best Regards Mikhail Lischuk -- _ -- 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] Operator panel with email sending capabilty
Hello, For an operator, I'm looking for a software application with which operator would be both able: - to see the list of awaiting calls, - to fill a (customizable) form with the name, number and reasonto use whern returning the call. Suggestions ? 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
Re: [asterisk-users] Questions about extension.conf
On 29/11/2012 11:47 AM, Salman Zafar wrote:
It is self explanatory, for example:
exten = _X.,1, Noop(Let say we have allowed all numbers i.e. _X
means and . specifies any range)
same = n,NoOp(Here we have skipped mentioning dial-pattern again and
thats it)
Hope I have answered your question.
Not for me.
What part of those lines and comments discusses same?
What is the syntax for a same line? what does it mean to use same
rather than exten?
On Thu, Nov 29, 2012 at 8:40 AM, Shitian Long longst...@gmail.com
mailto:longst...@gmail.com wrote:
Hello
I have been reading the sample extension.conf
;###
[outbound-freenum2]
; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
;
exten = _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same = n,Set(SUFFIX=${CUT(EXTEN,*,2-)}) ;
make sure the suffix is all digits as well
same = n,GotoIf($[${FILTER(0-9,${SUFFIX})} !=
${SUFFIX}]?fn-CONGESTION,1)
; filter out bad characters per the
README-SERIOUSLY.best-practices.txt document
same = n,Set(TIMEOUT(absolute)=10800)
same = n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org
http://freenum.org)}) ; perform our lookup with freenum.org
http://freenum.org
same = n,GotoIf($[${isnresult} != ]?from)
same = n,Set(DIALSTATUS=CONGESTION)
same = n,Goto(fn-CONGESTION,1)
same = n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} = ]?dial)
; check if we set the FREENUMDOMAIN global variable in [global]
same = n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ;
if we did set it, then we'll use it for our outbound dialing domain
same = n(dial),Dial(SIP/${isnresult},40)
same = n,Goto(fn-${DIALSTATUS},1)
exten = fn-BUSY,1,Busy()
exten = _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same = n,Congestion()
;##
According to
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf;
Syntax for defining a context: keywords *exten*, *include*,
*ignorepat* and *switch*. same is not mentioned in this wiki.
There is a part of dial plan from sample extension.conf above. My
Question is how same = key word works .
Thanks
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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--
Regards
**
Muhammad Salman
***
--
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--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102
--
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Re: [asterisk-users] Questions about extension.conf
That is a good answer. Thanks. Any reason why it is not documented? Ron On 29/11/2012 11:52 AM, Mikhail Lischuk wrote: Shitian Long wrote 29.11.2012 18:40: There is a part of dial plan from sample extension.conf above. My Question is how same = key word works . Thanks same is used for complex templates, if you don't want to copy previous line or afraid you can make a typo. exten = _1XXNXXX,1,Answer same = n,HangUp is the substitution for: exten = _1XXNXXX,1,Answer exten = _1XXNXXX,n,HangUp Also, it makes grepping the particular exten in a file a lot easier, and if you want to change some template for exten which has 50 lines, you don't have to edit all 50 of them. -- With Best Regards Mikhail Lischuk mailto:mlisc...@itx.com.ua -- _ -- 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 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Questions about extension.conf
The Wiki is (always) out of date. You might consider taking a look at
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#DialplanBasics_id262049
which is likely less out of data.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Thursday, November 29, 2012 12:17 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Questions about extension.conf
On 29/11/2012 11:47 AM, Salman Zafar wrote:
It is self explanatory, for example:
exten = _X.,1, Noop(Let say we have allowed all numbers i.e. _X
means and . specifies any range)
same = n,NoOp(Here we have skipped mentioning dial-pattern again and
thats it)
Hope I have answered your question.
Not for me.
What part of those lines and comments discusses same?
What is the syntax for a same line? what does it mean to use same rather
than exten?
On Thu, Nov 29, 2012 at 8:40 AM, Shitian Long longst...@gmail.com
wrote:
Hello
I have been reading the sample extension.conf
;###
[outbound-freenum2]
; This is the handler which performs the dialing logic. It is
called
; from the [outbound-freenum] context
;
exten = _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same = n,Set(SUFFIX=${CUT(EXTEN,*,2-)})
; make sure the suffix is all digits as well
same = n,GotoIf($[${FILTER(0-9,${SUFFIX})} !=
${SUFFIX}]?fn-CONGESTION,1)
; filter out bad characters per the README-SERIOUSLY.best-practices.txt
document
same = n,Set(TIMEOUT(absolute)=10800)
same =
n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ; perform our
lookup with freenum.org
same = n,GotoIf($[${isnresult} != ]?from)
same = n,Set(DIALSTATUS=CONGESTION)
same = n,Goto(fn-CONGESTION,1)
same = n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} = ]?dial)
; check if we set the FREENUMDOMAIN global variable in [global]
same = n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
;if we did set it, then we'll use it for our outbound dialing domain
same = n(dial),Dial(SIP/${isnresult},40)
same = n,Goto(fn-${DIALSTATUS},1)
exten = fn-BUSY,1,Busy()
exten = _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same = n,Congestion()
;##
According to
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf;
Syntax for defining a context: keywords exten, include,
ignorepat and switch. same is not mentioned in this wiki.
There is a part of dial plan from sample extension.conf above.
My Question is how same = key word works .
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
--
Regards
**
Muhammad Salman
***
--
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asterisk-users mailing list
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--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102
--
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Re: [asterisk-users] Questions about extension.conf
On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote: That is a good answer. Thanks. Any reason why it is not documented? It's documented on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Contexts,+Extensions,+and+Priorities Ron -- David M. Lee Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Operator panel with email sending capabilty
FOP (Flash Operator Panel) can probably do this for you. Personally I would do this with Perl, but other posters prefer C or PHP for this type of roll your own function. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Thursday, November 29, 2012 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Operator panel with email sending capabilty Hello, For an operator, I'm looking for a software application with which operator would be both able: - to see the list of awaiting calls, - to fill a (customizable) form with the name, number and reasonto use whern returning the call. Suggestions ? 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
Re: [asterisk-users] Questions about extension.conf
Excellent. It appears that Getting Started has a lot more stuff in it than the documentation for 1.8. Very helpful. Ron On 29/11/2012 12:31 PM, David M. Lee wrote: On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote: That is a good answer. Thanks. Any reason why it is not documented? It's documented on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Contexts,+Extensions,+and+Priorities Ron -- David M. Lee Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com http://www.digium.com/ www.asterisk.org http://www.asterisk.org/ -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Need help designing implementation
Hi, I'd like to replace my current VOIP provider with an Asterisk based solution. I have some ideas I want to run by the list to see if they are possible, and get answers to a couple questions. I want to setup two Asterisk servers that are linked to each other: - The first server would be my external (public) server and would live in a real data center. The second server would be my internal (private) server and would live in my house. - The external server would receive all incoming calls and handle the voice mail stuff. - The internal server would run all the phones in my house (VOIP or Analog-via-FXS). All outgoing calls would be routed out through the external server. I also want to add the following additional functionality: - If the external server looses connectivity to the internal server while a call is in progress, the external server should place the call on hold while it tries to reach us via our cell phones. A message should be played informing the remote party that the connection had been lost and it is trying to re-establish it now. If it can't reach us, it should inform the remote party that the connection could not be re-established and allow the remote party to leave some closing remarks on the voice mail system. - If a call comes in and no one is at home to take the call (or if all lines at home are busy), it should ring all of our cell phones and whoever answers the call first gets the call. If no one answers the call via the cell phones after 3 rings, it should route the call to the voice mail system. I say 3 rings on the cell phone because I do not want the cell phone voice mail to take the call. - I also would like the system to automatically route all calls directly to voice mail depending on the time of day (say 10PM to 8AM). I would like specify in a white list specific phone numbers that are allowed to ring through regardless of time of day (i.e. her parents, my parents). - I would like the VOIP phones to turn on the voice mail waiting indicator light if the external server has new voice messages. Is all of this possible? If not, which part's are not (and how much work do you think would be needed to make those parts work)? -- Thanks, Dyweni -- _ -- 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] Need help designing implementation
On 29/11/12 6:33 pm, Dyweni - Asterisk-Users wrote: I want to setup two Asterisk servers that are linked to each other: - The first server would be my external (public) server and would live in a real data center. The second server would be my internal (private) server and would live in my house. - The external server would receive all incoming calls and handle the voice mail stuff. - The internal server would run all the phones in my house (VOIP or Analog-via-FXS). All outgoing calls would be routed out through the external server. That all seems perfectly doable. - If the external server looses connectivity to the internal server while a call is in progress, the external server should place the call on hold while it tries to reach us via our cell phones. A message should be played informing the remote party that the connection had been lost and it is trying to re-establish it now. If it can't reach us, it should inform the remote party that the connection could not be re-established and allow the remote party to leave some closing remarks on the voice mail system. I don't think that's doable without quite a lot of work - but others may be able to advise further. To elaborate a little, it's easy to detect whether a route is usable when a call is placed, but detecting a call failure *during* the call is much more difficult. - If a call comes in and no one is at home to take the call (or if all lines at home are busy), it should ring all of our cell phones and whoever answers the call first gets the call. If no one answers the call via the cell phones after 3 rings, it should route the call to the voice mail system. I say 3 rings on the cell phone because I do not want the cell phone voice mail to take the call. That's easy, though remember asterisk does things in seconds rather than rings. You should also remember there's a delay in processing the call through the mobile networks before the phone actually starts ringing - in the UK that averages around 7 seconds between the call being sent to the mobile network from your server, and the phone ringing. - I also would like the system to automatically route all calls directly to voice mail depending on the time of day (say 10PM to 8AM). I would like specify in a white list specific phone numbers that are allowed to ring through regardless of time of day (i.e. her parents, my parents). Shouldn't be difficult. - I would like the VOIP phones to turn on the voice mail waiting indicator light if the external server has new voice messages. I believe this is doable in the newer versions of asterisk, but not the older versions. Again, someone else will hopefully chip in here, since our stuff is still running 1.4 :-) Is all of this possible? If not, which part's are not (and how much work do you think would be needed to make those parts work)? As is so often the case, (almost) anything is possible if you're prepared to spend time doing it. How much is worth doing depends on your time, and what else you might prefer to be doing with it... FWIW, you might want to think about whether you actually need a separate asterisk box at home. In my experience, unless you have many dozens of extensions, you're almost better off (and certainly no worse off) connecting your SIP devices at home (assuming you're using SIP) directly back to the * server in the datacentre. One less box to maintain, and things like MWI will just work without having to play with the messaging interfaces. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Scheduled Maintenance for Asterisk Project community services
On Friday, November 30th, 2012, the Asterisk community services listed below will be undergoing maintenance (migration to a new server and software upgrades). The services will be shut down at approximately 10:30 AM CST (4:30 PM December 1st UTC), and should return no later than 11:30 AM CST. Please keep in mind that it may take longer for our DNS updates to propagate throughout the Internet. We apologize in advance for any inconvenience this may cause. The affected services are: git.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
[asterisk-users] Scheduled Maintenance for Asterisk Project community services
On Friday, November 30th, 2012, the Asterisk community services listed below will be undergoing maintenance (migration to a new server and software upgrades). The services will be shut down at approximately 10:30 AM CST (4:30 PM December 1st UTC), and should return no later than 11:30 AM CST. Please keep in mind that it may take longer for our DNS updates to propagate throughout the Internet. We apologize in advance for any inconvenience this may cause. The affected services are: git.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
[asterisk-users] Need qualifications of SIP trunk providers
Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Thanks in advance! Elder D. Arohuanca dCAP 1497 Lima - Peru -- _ -- 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] Need qualifications of SIP trunk providers
On Thu, Nov 29, 2012 at 3:22 PM, Daniel - Asterisk earohua...@gmail.comwrote: Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Vitelity is reliable and decent, but no phone support. Have not used the others. Oh also if you lose a number on Vitelity to a port-out, they won't know and won't stop billing you for it. What's your expected volume in/out? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Need qualifications of SIP trunk providers
Several in our group use voip.ms and have no complaints at all We had a few hickups when Sandy rolled through NYC ( we are all on the NYC server ) but voip.ms responded quickly to mirror to Seattle and there was little downtime, and what was lasted a very short time on one day. voip.ms was very responsive during this time We all also use the IAX protocol supported by voip.ms and have no complaints John Novack Daniel - Asterisk wrote: Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Thanks in advance! Elder D. Arohuanca dCAP 1497 Lima - Peru -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] Need qualifications of SIP trunk providers
At 02:22 PM 11/29/2012, you wrote: Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms I started using Flowroute in Jan 2009 and have been very happy with their service. I'm small though, $5 to $20 / month with 2 numbers. On the very few times I've called with problems, mine or theirs, they've always been both helpful and knowledgeable, more than I might expect for someone my size. Ira -- _ -- 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] audio trouble with asterisk, help very much appreciated
From: Jody Gugelhupf knuef...@yahoo.com To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Sent: Thursday, November 29, 2012 10:23:01 PM Subject: audio trouble with asterisk, help very much appreciated Hi there :) first about my setup, running centos 6.2, asterisk 1.8.13.0, freepbx 2.9.0.12. I have a modem/router with NAT enabled. Asterisk and my extension from which I make the call are on the same local network, behind the modem/router. I have forwarded ports 1-2 to asterisk and configured asterisk accordingly. My external IP is 195.205.34.24. Asterisk and freepbx are on 192.168.2.199, my extension I make the calls from is '903' and is on 192.168.2.202. I have configured my extension and connected it. Also I have setup sip trunks and configured outbound rules etc. This all works fine. When I receive calls, all works great, I have two way audio without any trouble. When I make an outbound call, the incoming audio works without flaws, however my outgoing audio drops after a minute or so. So first the other person can hear me then not. On some calls the outgoing audio starts working again after a bit, but then drops again. In freepbx in the 'asterisk sip settings' I'm not sure how to set the NAT settings properly. Currently: NAT: yes (but have also tried 'no', 'never', and 'route' whilst keeping same settings below with same audio problems results as described above) IP Configuration: Static IP External IP: 195.205.34.24 Local Networks: 192.168.2.0/255.255.255.0 Would this be correct? I have run 'sip set debug on' and 'rtp set debug on' to see what happens during the call, the output is here: http://pastebin.ca/2257975 Would appreciate any help. Thank you in advance! :) Jody-- _ -- 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 number type always changed from subscriber user to national in libpri
hello:I used libpri 1.4.12 version with asterisk 1.8.7, after the pcap files, i found thatin wireshark setup message, the number type always changed from subscriber to national number.i have set pridialplan= local and prilocaldialplan=local in chan_dahdi.conf already. because that, the systemsometimes can not make outgoing calls. anyone can clarify that? Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.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
