Reuben Fine
2011-Sep-28 23:52 UTC
[asterisk-users] res_ODBC and failover (Bryant Zimmerman)
Hi Bryant please use SRV records whenever a failover is required. This your
hosting provider should be able to give to you and always try and avoid using
direct ip calls.
Potentially you could create two A records:
db1.yoursite.com
db2.yoursite.com
Then create a SRV record with priority on db1 over db2. This way you can add
n-amount of additional failovers.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com
Sent: Wednesday, September 28, 2011 6:01 PM
To: asterisk-users at lists.digium.com
Subject: asterisk-users Digest, Vol 86, Issue 57
Send asterisk-users mailing list submissions to
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When replying, please edit your Subject line so it is more specific than
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Today's Topics:
1. Re: Receiving musinc on hold instead of ring (Alejandro Recarey)
2. Re: Receiving musinc on hold instead of ring (Tarek Sawah)
3. Limit outbond calls duration to 1 minute (salaheddine elharit)
4. FreeTDS and MS-SQL with Asterisk RealTime (Reuben Fine)
5. Re: Limit outbond calls duration to 1 minute (Paul Belanger)
6. Re: Limit outbond calls duration to 1 minute (Tarek Sawah)
7. Re: Limit outbond calls duration to 1 minute (salaheddine elharit)
8. Re: Limit outbond calls duration to 1 minute (salaheddine elharit)
9. Re: Limit outbond calls duration to 1 minute (Danny Nicholas)
10. Re: Limit outbond calls duration to 1 minute (Tarek Sawah)
11. Re: Limit outbond calls duration to 1 minute (salaheddine elharit)
12. Re: Limit outbond calls duration to 1 minute (Tarek Sawah)
13. res_ODBC and failover (Bryant Zimmerman)
14. Anybody using BinFone Telecom? (ftarz at mindspring.com)
15. Increasing the fxorxgain and fxotxgain for the hardware of
the digium card (bilal ghayyad)
----------------------------------------------------------------------
Message: 1
Date: Wed, 28 Sep 2011 18:59:39 +0200
From: Alejandro Recarey <alexrecarey at gmail.com>
Subject: Re: [asterisk-users] Receiving musinc on hold instead of ring
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<CA+zbGr8ARS-E2m+wvmM4BVGwUknBuoVdvamgBC0QR4XM2q8j0Q at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
> this is related to your carrier's SIP messages as they are sending a
> sendonly attribute instead of sendrecv (taking a wild guess here) your
> asterisk will act as if the call was placed on hold thus the MOH butts in.
> an sip debug log for a similar call will be more helpful?
Thanks for the answer Tarek! I will try to obtain a full SIP trace tonight. If
the problem is indeed that the carrier is sending the sendonly attribute in the
SDP instead of sendrecv, what can I do? Is there anything I can configure on my
side?
Thanks again,
Alex
------------------------------
Message: 2
Date: Wed, 28 Sep 2011 17:34:11 +0000
From: Tarek Sawah <tareksawah at hotmail.com>
Subject: Re: [asterisk-users] Receiving musinc on hold instead of ring
To: Asterisk Users <asterisk-users at lists.digium.com>
Message-ID: <SNT121-W34E19228122B3A2ACA6E1DCAF10 at phx.gbl>
Content-Type: text/plain; charset="iso-8859-1"
i have faced this problem with one of the major VoIP whole providers in India
.. they have a new platform with Sonus switches.. which does not support
sendrecv media attribute .. however a work around that may work for you .. is
enabling re-invite on their peer.
let me know if this works out for you.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
> From: alexrecarey at gmail.com
> Date: Wed, 28 Sep 2011 18:59:39 +0200
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Receiving musinc on hold instead of ring
>
> > this is related to your carrier's SIP messages as they are sending
a
> > sendonly attribute instead of sendrecv (taking a wild guess here)
> > your asterisk will act as if the call was placed on hold thus the MOH
butts in.
> > an sip debug log for a similar call will be more helpful?
>
> Thanks for the answer Tarek! I will try to obtain a full SIP trace
> tonight. If the problem is indeed that the carrier is sending the
> sendonly attribute in the SDP instead of sendrecv, what can I do? Is
> there anything I can configure on my side?
>
> Thanks again,
>
> Alex
>
> --
> _____________________________________________________________________
> -- 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
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Message: 3
Date: Wed, 28 Sep 2011 17:59:27 +0000
From: salaheddine elharit <salah.elharit200 at gmail.com>
Subject: [asterisk-users] Limit outbond calls duration to 1 minute
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<CAHexamvgcV8FEV2pLjEiGqCEACoHf7gNyDyBQ07TN3a_Z9gLeA at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
hello list
i have configured a sip account in order to do an outbound calls and i want to
force a hang up after 1 min for 222 sip
in extensions.conf i have
exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => 222,n,AbsoluteTimeout(60)
exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten => 222,n,Dial(SIP/${EXTEN},,KkTt)
exten => 222,n,Hangup();
could you please see this code and tell me waht is wrong
thanks and regards
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------------------------------
Message: 4
Date: Wed, 28 Sep 2011 18:01:35 +0000
From: Reuben Fine <fine at allmysons.com>
Subject: [asterisk-users] FreeTDS and MS-SQL with Asterisk RealTime
To: "asterisk-users at lists.digium.com"
<asterisk-users at lists.digium.com>
Message-ID:
<8BB5D54201FF7A45AB99A14D0C37C1660773D7 at
CH1PRD0702MB107.namprd07.prod.outlook.com>
Content-Type: text/plain; charset="us-ascii"
We have successfully setup and tested integration between Asterisk and MS-SQL.
We are currently running about 70 simultaneous calls throughout the day however
after some time our MS-SQL server (Windows 2008 64bit, SQL Server 2008) starts
to increase it's memory usage exponentially. The MS-SQL server CPU also pegs
at around 90%+ and becomes unresponsive and cannot accept new connections. We
are running Asterisk 1.8.6 currently. FreeTDS version is 4.2 and UnixODBC is
2.2.12. The kernel information is : Linux 2.6.27.25-78.2.56.fc9.i686.PAE #1 SMP
Thu Jun 18 12:36:07 EDT 2009 i686 i686 i386 GNU/Linux. We are using Realtime and
using FreeTDS to connect to the MS-SQL server where we control sip users,
voicemail and so forth. This works fine however when we enable CEL and CDR into
MS-SQL the server begins to grow in memory usage / CPU usage until the SQL
server halts and stops taking new requests.
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Message: 5
Date: Wed, 28 Sep 2011 14:02:04 -0400
From: Paul Belanger <pabelanger at digium.com>
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <4E83611C.1010902 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
On 11-09-28 01:59 PM, salaheddine elharit wrote:> hello list
>
>
> i have configured a sip account in order to do an outbound calls and i
> want to force a hang up after 1 min for 222 sip
>
>
> in extensions.conf i have
>
>
> exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
>
> exten => 222,n,AbsoluteTimeout(60)
>
>
> exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
>
> exten => 222,n,Dial(SIP/${EXTEN},,KkTt)
>
> exten => 222,n,Hangup();
>
> could you please see this code and tell me waht is wrong
>
*CLI> core show application Dial
Look at the 'L' flag
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at:
http://digium.com & http://asterisk.org
------------------------------
Message: 6
Date: Wed, 28 Sep 2011 18:08:59 +0000
From: Tarek Sawah <tareksawah at hotmail.com>
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
To: Asterisk Users <asterisk-users at lists.digium.com>
Message-ID: <SNT121-W40B9D1E407CFD99F3A2158CAF10 at phx.gbl>
Content-Type: text/plain; charset="iso-8859-1"
have a look at the following:
"L(x[:y][:z]): Limit the call to 'x'
ms, warning when 'y' ms are left, repeated every 'z' ms) Only
'x' is
required, 'y' and 'z' are optional."
source
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 28 Sep 2011 17:59:27 +0000
From: salah.elharit200 at gmail.com
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Limit outbond calls duration to 1 minute
hello list
i have configured a sip account in order to do an outbound calls and i want to
force a hang up after 1 min for 222 sip
in extensions.conf i have
exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => 222,n,AbsoluteTimeout(60)
exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten => 222,n,Dial(SIP/${EXTEN},,KkTt)
exten => 222,n,Hangup();
could you please see this code and tell me waht is wrong
thanks and regards
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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Message: 7
Date: Wed, 28 Sep 2011 18:09:32 +0000
From: salaheddine elharit <salah.elharit200 at gmail.com>
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<CAHexamu+cno_phduHXiXsTiB1oYEFWy-MFjFgYFOLDy6YHX77w at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
i have this when
L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when
'y' ms are
left. Repeat the warning every 'z' ms. The following special
variables can be used with this option:
* LIMIT_PLAYAUDIO_CALLER yes|no (default yes)
Play sounds to the caller.
* LIMIT_PLAYAUDIO_CALLEE yes|no
Play sounds to the callee.
* LIMIT_TIMEOUT_FILE File to play when time is up.
* LIMIT_CONNECT_FILE File to play when call begins.
* LIMIT_WARNING_FILE File to play as warning if 'y' is
defined.
The default is to say the time
remaining.
but i don't understand what i can do to solve this
thanks
2011/9/28 Paul Belanger <pabelanger at digium.com>
> On 11-09-28 01:59 PM, salaheddine elharit wrote:
>
>> hello list
>>
>>
>> i have configured a sip account in order to do an outbound calls and i
>> want
>> to force a hang up after 1 min for 222 sip
>>
>>
>> in extensions.conf i have
>>
>>
>> exten => 222,1,MixMonitor(sip_${EXTEN}_**${UNIQUEID}.wav|av(0}V(0))
>>
>> exten => 222,n,AbsoluteTimeout(60)
>>
>>
>> exten => 222,n,Set(AUDIOHOOK_INHERIT(**MixMonitor)=yes)
>>
>> exten => 222,n,Dial(SIP/${EXTEN},,KkTt)
>>
>> exten => 222,n,Hangup();
>>
>> could you please see this code and tell me waht is wrong
>>
>> *CLI> core show application Dial
>
> Look at the 'L' flag
>
> --
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://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<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
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Message: 8
Date: Wed, 28 Sep 2011 18:22:57 +0000
From: salaheddine elharit <salah.elharit200 at gmail.com>
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<CAHexamuXv0Awg1nDP+4-8t4X6=Zf9qEvGH3aN=Y8XEMgJWqT1Q at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
but there is no exemple for when i must put X in order to limit the call
can you please give me an exemple
regards
2011/9/28 Tarek Sawah <tareksawah at hotmail.com>
> have a look at the following:
> "*L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when
'y' ms are left,
> repeated every 'z' ms) Only 'x' is required, 'y'
and 'z' are optional."
>
>
> source
> http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
>
> Tarek Sawah
>
> Information Technology Adviser
>
> Integrated Digital Systems
>
> CCNP, MCSE, RHCE, TELECOM
>
> USA: +1 386 492 9993
>
>
>
> ------------------------------
> Date: Wed, 28 Sep 2011 17:59:27 +0000
> From: salah.elharit200 at gmail.com
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] Limit outbond calls duration to 1 minute
>
>
> hello list
>
>
> i have configured a sip account in order to do an outbound calls and i want
> to force a hang up after 1 min for 222 sip
>
>
> in extensions.conf i have
>
> exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
> exten => 222,n,AbsoluteTimeout(60)
>
> exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
> exten => 222,n,Dial(SIP/${EXTEN},,KkTt)
> exten => 222,n,Hangup();
> could you please see this code and tell me waht is wrong
> thanks and regards
>
>
>
> -- _____________________________________________________________________ --
> Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
> Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE
> or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Message: 9
Date: Wed, 28 Sep 2011 13:25:14 -0500
From: "Danny Nicholas" <danny at debsinc.com>
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <013d01cc7e0b$f75ae110$e610a330$@debsinc.com>
Content-Type: text/plain; charset="us-ascii"
As I read this, the following should be correct:
exten => 222,n,Dial(SIP/${EXTEN},,KkTtL(60000))
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Wednesday, September 28, 2011 1:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
but there is no exemple for when i must put X in order to limit the call
can you please give me an exemple
regards
2011/9/28 Tarek Sawah <tareksawah at hotmail.com>
have a look at the following:
"L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y'
ms are left,
repeated every 'z' ms) Only 'x' is required, 'y' and
'z' are optional."
source
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
_____
Date: Wed, 28 Sep 2011 17:59:27 +0000
From: salah.elharit200 at gmail.com
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Limit outbond calls duration to 1 minute
hello list
i have configured a sip account in order to do an outbound calls and i want
to force a hang up after 1 min for 222 sip
in extensions.conf i have
exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => 222,n,AbsoluteTimeout(60)
exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten => 222,n,Dial(SIP/${EXTEN},,KkTt)
exten => 222,n,Hangup();
could you please see this code and tell me waht is wrong
thanks and regards
-- _____________________________________________________________________ --
Bandwidth and Colocation Provided by http://www.api-digital.com
<http://www.api-digital.com/> -- New to Asterisk? Join us for a live
introductory webinar every Thurs: http://www.asterisk.org/hello
asterisk-users mailing list To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com
<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:
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Message: 10
Date: Wed, 28 Sep 2011 18:31:50 +0000
From: Tarek Sawah <tareksawah at hotmail.com>
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
To: Asterisk Users <asterisk-users at lists.digium.com>
Message-ID: <SNT121-W2278C3A876143D267F4929CAF10 at phx.gbl>
Content-Type: text/plain; charset="iso-8859-1"
exten => 222,n,Dial(SIP/${EXTEN},,KkTtLL(60000:30000:10000))
this will call the extension and sets the limit to 60000MS which equals 60
seconds.. and will inform the caller of his remaining time when he has only 30
seconds left.. and will repeat the notification every ten seconds (this is an
over do and playing such sounds files at this rate will consume the resources!)
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 28 Sep 2011 18:22:57 +0000
From: salah.elharit200 at gmail.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
but there is no exemple for when i must put X in order to limit the call
can you please give me an exemple
regards
2011/9/28 Tarek Sawah <tareksawah at hotmail.com>
have a look at the following:
"L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y'
ms are left, repeated every 'z' ms) Only 'x' is required,
'y' and 'z' are optional."
source
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 28 Sep 2011 17:59:27 +0000
From: salah.elharit200 at gmail.com
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Limit outbond calls duration to 1 minute
hello list
i have configured a sip account in order to do an outbound calls and i want to
force a hang up after 1 min for 222 sip
in extensions.conf i have
exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => 222,n,AbsoluteTimeout(60)
exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten => 222,n,Dial(SIP/${EXTEN},,KkTt)
exten => 222,n,Hangup();
could you please see this code and tell me waht is wrong
thanks and regards
-- _____________________________________________________________________ --
Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or
update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
_____________________________________________________________________
-- 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 UNSUBSCRIBE or update options visit:
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------------------------------
Message: 11
Date: Wed, 28 Sep 2011 18:32:28 +0000
From: salaheddine elharit <salah.elharit200 at gmail.com>
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<CAHexamudxVkYZTRP4jC4r1o+UbN7KVLGU46VY77m4xV5R3ZR_w at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
sorry but the issue still the same there is no hangup after 1Min
regards
2011/9/28 Danny Nicholas <danny at debsinc.com>
> As I read this, the following should be correct:****
>
> exten => 222,n,Dial(SIP/${EXTEN},,KkTtL(60000))
>
> ****
>
> ** **
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Wednesday, September 28, 2011 1:23 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Limit outbond calls duration to 1 minute**
> **
>
> ** **
>
> but there is no exemple for when i must put X in order to limit the call**
> **
>
> ****
>
> can you please give me an exemple****
>
> ****
>
> regards****
>
> 2011/9/28 Tarek Sawah <tareksawah at hotmail.com>****
>
> have a look at the following:
> "*L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when
'y' ms are left,
> repeated every 'z' ms) Only 'x' is required, 'y'
and 'z' are optional."
>
>
> source
> http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
>
> Tarek Sawah
>
> Information Technology Adviser
>
> Integrated Digital Systems
>
> CCNP, MCSE, RHCE, TELECOM
>
> USA: +1 386 492 9993
>
>
> ****
> ------------------------------
>
> Date: Wed, 28 Sep 2011 17:59:27 +0000
> From: salah.elharit200 at gmail.com
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] Limit outbond calls duration to 1 minute ****
>
> ** **
>
> hello list ****
>
> ****
>
> i have configured a sip account in order to do an outbound calls and i want
> to force a hang up after 1 min for 222 sip****
>
> ****
>
> ****
>
> in extensions.conf i have ****
>
> ****
>
> exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
> exten => 222,n,AbsoluteTimeout(60)
>
> exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
> exten => 222,n,Dial(SIP/${EXTEN},,KkTt)
> exten => 222,n,Hangup();
> could you please see this code and tell me waht is wrong
> thanks and regards****
>
> ****
>
> ****
>
> ** **
>
> -- _____________________________________________________________________ --
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> or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users****
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> http://lists.digium.com/mailman/listinfo/asterisk-users****
>
> ** **
>
> --
> _____________________________________________________________________
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Message: 12
Date: Wed, 28 Sep 2011 18:37:15 +0000
From: Tarek Sawah <tareksawah at hotmail.com>
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
To: Asterisk Users <asterisk-users at lists.digium.com>
Message-ID: <SNT121-W5177D57EFC4CCC2C0B51DCAF10 at phx.gbl>
Content-Type: text/plain; charset="iso-8859-1"
one adjustment i would suggest is using (|) instead of (,)
exten => 222,n,Dial(SIP/${EXTEN}||KkTtL(60000))
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 28 Sep 2011 18:32:28 +0000
From: salah.elharit200 at gmail.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
sorry but the issue still the same there is no hangup after 1Min
regards
2011/9/28 Danny Nicholas <danny at debsinc.com>
As I read this, the following should be correct:
exten => 222,n,Dial(SIP/${EXTEN},,KkTtL(60000))
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of salaheddine elharit
Sent: Wednesday, September 28, 2011 1:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
but there is no exemple for when i must put X in order to limit the call
can you please give me an exemple
regards
2011/9/28 Tarek Sawah <tareksawah at hotmail.com>
have a look at the following:
"L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y'
ms are left, repeated every 'z' ms) Only 'x' is required,
'y' and 'z' are optional."
source
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 28 Sep 2011 17:59:27 +0000
From: salah.elharit200 at gmail.com
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Limit outbond calls duration to 1 minute
hello list
i have configured a sip account in order to do an outbound calls and i want to
force a hang up after 1 min for 222 sip
in extensions.conf i have
exten => 222,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => 222,n,AbsoluteTimeout(60)
exten => 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten => 222,n,Dial(SIP/${EXTEN},,KkTt)
exten => 222,n,Hangup();
could you please see this code and tell me waht is wrong
thanks and regards
-- _____________________________________________________________________ --
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Message: 13
Date: Wed, 28 Sep 2011 16:30:37 -0400
From: "Bryant Zimmerman" <BryantZ at zktech.com>
Subject: [asterisk-users] res_ODBC and failover
To: <asterisk-users at lists.digium.com>
Message-ID: <3689ea1e$37cd9d4c$1d6777e$@com>
Content-Type: text/plain; charset="us-ascii"
I am toying with res_ODBC. currently I am using dns=ODBCvalue. Is there a
way to fail this over to another dns value in the event the a primary is
off line.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
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Message: 14
Date: Wed, 28 Sep 2011 18:33:49 -0400
From: "ftarz at mindspring.com" <ftarz at mindspring.com>
Subject: [asterisk-users] Anybody using BinFone Telecom?
To: asterisk-users at lists.digium.com
Message-ID: <4E83A0CD.7070001 at mindspring.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Does anyone have any experience with BinFone for IAX termination?
They good look on the website, but I'm looking for any comments.
------------------------------
Message: 15
Date: Wed, 28 Sep 2011 15:59:31 -0700 (PDT)
From: bilal ghayyad <bilmar_gh at yahoo.com>
Subject: [asterisk-users] Increasing the fxorxgain and fxotxgain for
the hardware of the digium card
To: asterisk-users at lists.digium.com
Message-ID:
<1317250771.98667.YahooMailClassic at web162014.mail.bf1.yahoo.com>
Content-Type: text/plain; charset=us-ascii
Hi All;
In the zaptel, we were increasing the gain of the voice volume at the hardware
level from the /etc/zaptel and /etc/modprob.conf files, but now we are using
DAHDI, so where to do the same thing?
I am looking actually to increase the volume at hardware level and not software
to avoid the DTMF detection problem and to have better voice quality.
Any advise?
Regards
Bilal
------------------------------
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End of asterisk-users Digest, Vol 86, Issue 57
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