[asterisk-users] PSTN connectivity

2011-09-28 Thread michael k 
Hi All,

  I am trying to connect my asterisk box with freepbx to PSTN. I
have purchased x100p FXO card and installed in my asterisk server. My
freepbx detected the x100p FXO card and i can see the card specific details
in command line. I have configured the following things.

1. OUTBOUND caller id and Dialing rules in Freepbx.

2. INBOUND route

When i call to the PSTN number before connecting to the FXO card, i am
getting a ringing. But i get a message like the "number is out of order"
when i just connect the line to FXO card.

Please some one help me to resolve his issue
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Re: [asterisk-users] PSTN connectivity

2011-09-28 Thread Sam Govind 
Some CLI logs will get you better help on the issue ! also paste the FXO
configurations and how you configured it !

On Wed, Sep 28, 2011 at 2:11 PM, michael k  wrote:

> Hi All,
>
>   I am trying to connect my asterisk box with freepbx to PSTN. I
> have purchased x100p FXO card and installed in my asterisk server. My
> freepbx detected the x100p FXO card and i can see the card specific details
> in command line. I have configured the following things.
>
> 1. OUTBOUND caller id and Dialing rules in Freepbx.
>
> 2. INBOUND route
>
> When i call to the PSTN number before connecting to the FXO card, i am
> getting a ringing. But i get a message like the "number is out of order"
> when i just connect the line to FXO card.
>
> Please some one help me to resolve his issue
>
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Re: [asterisk-users] Asterisk Realtime Time Dial App

2011-09-28 Thread Nick Khamis 
Hello David,

I have this discussion also on the -dev mailing list. and suggested
that we use a database hook to trigger the originate process (pleasee
see "Outbound Call Implementation"). However, compiling it directly
into asterisk as a realtime moodule insted of using AMI etc...

Cheers,

Nick.



On Wed, Sep 28, 2011 at 1:23 AM, Sam Govind  wrote:
> Correct me if I'm wrong or don't know anything other than AMI Originate
> Event or a call file to kick start a call from asterisk ! So making a new or
> modifying asterisk call-file cron job/poller seems like a nice idea but why
> put on extra load on Asterisk. (See pbx_spool.c if still want to modify).
> The simple idea is create a MySQL trigger for your Table insertion, the data
> in the table at insertion time becomes parameters for a simple script that
> triggers an AMI event (or call file) whichever is easier for you.
>
> On Tue, Sep 27, 2011 at 6:54 PM, Nick Khamis  wrote:
>>
>> Hello David,
>>
>> At first I assumed asterisk used call files out of the box for
>> normal-initiated/instantiated calls however,
>> this is incorrect. I think call files was the easy approach for client
>> just to place a file with call details
>> in some location. I am trying to do the same with a db record. My
>> first question is, how does asterisk
>> initiate calls, i.e. what part of the source code is responsible for
>> that. Are there any threads involved etc.
>>
>> Cheers,
>>
>> Nick.
>>
>> On Tue, Sep 27, 2011 at 9:35 AM, David Moring 
>> wrote:
>> > Hi Nick,
>> >
>> > Understand your reasoning - though as Matt points out sql db isn't in
>> > the
>> > core so compiling it there would preclude seemless upgrades.  Also, I
>> > personally would be concerned putting the calls right into the call-file
>> > thread might create an issue if you hung on a db query or insert.
>> >  Finally
>> > (and I'd love to hear the answer not knowing), but I believe
>> > "normally-initiated/instantiated" calls are handled with direct calls
>> > via
>> > either SIP requests and/or AMI - thus even using the proposed method, I
>> > *think* the db/file-drop method is going to create some overhead that
>> > might
>> > not scale well...
>> >
>> > Best,
>> >
>> > David
>> >
>> > -Original Message-
>> > From: Nick Khamis 
>> > To: Asterisk Users Mailing List - Non-Commercial Discussion
>> > 
>> > Date: Mon, 26 Sep 2011 18:49:07 -0400
>> > Subject: Re: [asterisk-users] Asterisk Realtime Time Dial App
>> >
>> > Hello David,
>> >
>> > Thank you so much for your response. I am sure it can be easily done
>> > using AGI. The reason I am leaning more
>> > towards storing the call information in a database record, is because
>> > our existing client applications can be easily
>> > modified to write to MySQL. The asterisk cron/thread that would
>> > querying the DB should be no different than existing implementation
>> > used process the call files?
>> > For those of you that may be interested in what we are doing. We are
>> > developing an application that will apply NLP
>> > services on text generated using the speech to text module, and
>> > generate the response that will then be forwarded to
>> > the text to speech.
>> >
>> > Cheers,
>> >
>> > Nick
>> > .
>> >
>> >
>> >
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>> >
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Re: [asterisk-users] Receiving musinc on hold instead of ring

2011-09-28 Thread Tarek Sawah 

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?

Tarek Sawah

Information Technology  Adviser

Integrated Digital Systems

CCNP, MCSE, RHCE, TELECOM

USA: +1 386 492 9993



> From: alexreca...@gmail.com
> Date: Wed, 28 Sep 2011 03:44:35 +0200
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Receiving musinc on hold instead of ring
> 
> Hi all and thanks for reading.
> 
> I am having a very strange issue. When dialing out with a certain
> carrier, asterisk 1.6.20 will play music on hold instead of a ring
> tone, although this behaviour is NOT what I want.
> 
> Example dialplan execution:
> 
> -- Executing [0021266xxx@test:13] Progress("SIP/100-1e04", "") in new 
> stack
> -- Executing [0021266xxx@test:14]
> Dial("SIP/100-1e04","SIP/21266xxx@x.x.x.x") in new stack
> -- Called 21266xxx@x.x.x.x
> -- Call on SIP/x.x.x.x-1e05 placed on hold
> -- Started music on hold, class 'default', on SIP/100-1e04
> -- SIP/x.x.x.x-1e05 is making progress passing it to SIP/100-1e04
> 
> Now, a SIP packet capture shows no trace of the call being put on hold!
> 
> Sample wireshark capture for the same call:
> 
> x.x.x.x -> y.y.y.y SIP/SDP Request: INVITE sip:21266xxx@x.x.x.x, with
> session description
> y.y.y.y -> x.x.x.x SIP Status: 100 Giving a try
> y.y.y.y -> x.x.x.x SIP/SDP Status: 180 Ringing, with session description
> 
> And I get the music on hold instead of the ringtone. I have tried
> placing Progress() in front of Dial() but to no avail. I do not want
> to use the "r" option in Dial() because then I lose the destination
> ringtone in early media which is important to my customers.
> 
> Anybody had a similar issue? Any idea of what parameters I can try to
> tweak, as I am stumped...
> 
> Thanks!
> 
> Alex
> 
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[asterisk-users] Scheduled Maintenance for Asterisk Project community services

2011-09-28 Thread Asterisk Development Team 
On Thursday, September 29th, 2011, the Asterisk community services 
listed below will be undergoing maintenance (software upgrades and 
updates). The services will be shut down at approximately 7:00 PM CDT 
(12:00 AM September 30 UTC), and will return no later than 8:00 PM CDT. 
We apologize in advance for any inconvenience this may cause.


The affected services are:

code.asterisk.org
wiki.asterisk.org
issues.asterisk.org/jira

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Re: [asterisk-users] Set (MONITOR_FILENAME=.................) for queuing recording calls

2011-09-28 Thread Jim Dickenson 
I do not know when the recording actually starts but if it start when the agent 
answers the call then it might be possible to have the name set in an AGI that 
gets run when the agent answers call. If nothing else you can set a variable to 
the name you want to have the file have and rename it at end of call.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Sep 27, 2011, at 10:30 PM, Sam Govind wrote:

> :P I'd this very similar situation/ project Carl - and guess what. The 
> filename is created before the call actually hits QUEUE application so these 
> Queue variables are not populated by then so filename won't contain the Agent 
> Number.
> UNLESS you move the file after queue to a new filename containing the Agent 
> Number.
> 
> like ;
> 
> exten => whatever,n,SET(MONITOR_FILENAME=blah-blah)
> exten => whatever,n,Queue(${params}); Queue should contain option "c" to 
> continue in dialplan when callee hangup. Caller hangup case needs special 
> attention too
> exten => whatever,n,System(mv ${old-Filename} 
> ${old-Filename}-${MEMBERINTERFACE})
> 
> I guess this should do the job.
> 
> On Tue, Sep 27, 2011 at 8:30 PM, Carlos Chavez  
> wrote:
> On Tue, 2011-09-27 at 03:47 -0700, bilal ghayyad wrote:
> > Dears;
> >
> > I am facing now a problem in the recording the calls that coming via the 
> > queue, the problem that I am not able to make the filename contains the 
> > agent (for example its extension) who received the call.
> >
> > Actually by looking to the below settings, it is clear that the agent name 
> > (it the phone extension or it is sip username .. etc) will not be included 
> > in the filename.
> >
> > How can I include the agent name in the filename? Because in outboud it is 
> > easy as the ${CHANNEL} will contain the sip username of the IP Phone but in 
> > the outbound it will contain the DAHDI channel that the call came via it .. 
> > so How to inlude the sip username for the IP Phone of the agent that is 
> > going to get the call from the queue?
> >
> > exten => 
> > s,1,Set(MONITOR_FILENAME=${CHANNEL}${CALLERID(num)}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
> > exten => s,2,Queue(OrangeCMG,t,,,180)
> > exten => s,3,Macro(voicemail,SIP/reception)
> >
> > Regards
> > Bilal
> >
> >
> ; If set to yes, just prior to the caller being bridged with a queue
> member
> ; the following variables will be set
> ; MEMBERINTERFACE is the interface name (eg. Agent/1234)
> ; MEMBERNAME is the member name (eg. Joe Soap)
> ; MEMBERCALLS is the number of calls that interface has taken,
> ; MEMBERLASTCALL is the last time the member took a call.
> ; MEMBERPENALTY is the penalty of the member
> ; MEMBERDYNAMIC indicates if a member is dynamic or not
> ; MEMBERREALTIME indicates if a member is realtime or not
> ;
> ;setinterfacevar=no
> 
> Basically the variable ${MEMBERINTERFACE} will have the extension (if
> using dynamic members) or the agent number.
> 
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
> 
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[asterisk-users] Asterisk Realtime SIP : vmexten

2011-09-28 Thread Jonas Kellens 

Hello list,

is the field "vmexten" available when using SIP peers in a realtime 
Mysql-DB ?



Thanks.

Kind regards,

Jonas.
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Re: [asterisk-users] number of calls simultaneous from AMI

2011-09-28 Thread Adolphe Cher-Aime 
Make sure that you set *async *option to true. If not asterisk will  wait
for response on previous calls  before making any other calls.

Hope that will  help.

On Wed, Sep 28, 2011 at 12:17 AM, Sam Govind  wrote:

> If you can post any relevant code sections and CLI output for this then
> it'll be lot better to determine whats causing this. I never got any problem
> initiating as many call as u can say from AMI !
>
> On Tue, Sep 27, 2011 at 5:36 PM, Jerry Geis  wrote:
>
>>  I am starting up 4 calls over the AMI.
>> It "appears" as though the first 3 start up and go out right away.
>> The 4th call is delayed like 15 seconds.
>>
>> Any thoughts on why this fourth call might be getting delayed...
>>
>> Everything is working its just delayed.
>>
>> Jerry
>>
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>
>
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Re: [asterisk-users] Receiving musinc on hold instead of ring

2011-09-28 Thread Alejandro Recarey 
> 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

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Re: [asterisk-users] Receiving musinc on hold instead of ring

2011-09-28 Thread Tarek Sawah 

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: alexreca...@gmail.com
> Date: Wed, 28 Sep 2011 18:59:39 +0200
> To: asterisk-users@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
> 
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[asterisk-users] Limit outbond calls duration to 1 minute

2011-09-28 Thread salaheddine elharit 
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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[asterisk-users] FreeTDS and MS-SQL with Asterisk RealTime

2011-09-28 Thread Reuben Fine 
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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Re: [asterisk-users] Limit outbond calls duration to 1 minute

2011-09-28 Thread Paul Belanger 

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

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Re: [asterisk-users] Limit outbond calls duration to 1 minute

2011-09-28 Thread Tarek Sawah 

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 +
From: salah.elharit...@gmail.com
To: asterisk-users@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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Re: [asterisk-users] Limit outbond calls duration to 1 minute

2011-09-28 Thread salaheddine elharit 
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 

>  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
>
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Re: [asterisk-users] Limit outbond calls duration to 1 minute

2011-09-28 Thread salaheddine elharit 
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 

>  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 +
> From: salah.elharit...@gmail.com
> To: asterisk-users@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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Re: [asterisk-users] Limit outbond calls duration to 1 minute

2011-09-28 Thread Danny Nicholas 
As I read this, the following should be correct:

exten => 222,n,Dial(SIP/${EXTEN},,KkTtL(6))



 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@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 

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 +
From: salah.elharit...@gmail.com
To: asterisk-users@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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Re: [asterisk-users] Limit outbond calls duration to 1 minute

2011-09-28 Thread Tarek Sawah 


exten => 222,n,Dial(SIP/${EXTEN},,KkTtLL(6:3:1))

this will call the extension and sets the limit to 6MS 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 +
From: salah.elharit...@gmail.com
To: asterisk-users@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 



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 +
From: salah.elharit...@gmail.com
To: asterisk-users@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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Re: [asterisk-users] Limit outbond calls duration to 1 minute

2011-09-28 Thread salaheddine elharit 
sorry but the issue still the same there is no hangup after 1Min

regards

2011/9/28 Danny Nicholas 

>  As I read this, the following should be correct:
>
> exten => 222,n,Dial(SIP/${EXTEN},,KkTtL(6))
>
> 
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine
> elharit
> *Sent:* Wednesday, September 28, 2011 1:23 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Limit outbond calls duration to 1 minute**
> **
>
> ** **
>
> but there is no exemple for when i must put X in order to limit the call**
> **
>
>  
>
> can you please give me an exemple
>
>  
>
> regards
>
> 2011/9/28 Tarek Sawah 
>
> have a look at the following:
> "*L(*x[:y][:z]*)*: Limit the call to 'x' ms, warning when 'y' ms are left,
> repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional."
>
>
> source
> http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
>
> Tarek Sawah
>
> Information Technology  Adviser
>
> Integrated Digital Systems
>
> CCNP, MCSE, RHCE, TELECOM
>
> USA: +1 386 492 9993
>
>
> 
>  --
>
> Date: Wed, 28 Sep 2011 17:59:27 +
> From: salah.elharit...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Limit outbond calls duration to 1 minute 
>
> ** **
>
> hello list 
>
>  
>
> i have configured a sip account in order to do an outbound calls and i want
> to force a hang up after 1 min for 222 sip
>
>  
>
>  
>
> in extensions.conf i have 
>
>  
>
> exten => 222,1,MixMonitor(sip_${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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> Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Limit outbond calls duration to 1 minute

2011-09-28 Thread Tarek Sawah 

one adjustment i would suggest is using (|) instead of (,)

exten => 222,n,Dial(SIP/${EXTEN}||KkTtL(6))



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 +
From: salah.elharit...@gmail.com
To: asterisk-users@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 




As I read this, the following should be correct:
exten => 222,n,Dial(SIP/${EXTEN},,KkTtL(6))


 
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@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 


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 +
From: salah.elharit...@gmail.com
To: asterisk-users@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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[asterisk-users] res_ODBC and failover

2011-09-28 Thread Bryant Zimmerman 
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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[asterisk-users] Anybody using BinFone Telecom?

2011-09-28 Thread ft...@mindspring.com 

Does anyone have any experience with BinFone for IAX termination?

They good look on the website, but I'm looking for any comments.

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[asterisk-users] Increasing the fxorxgain and fxotxgain for the hardware of the digium card

2011-09-28 Thread bilal ghayyad 
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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[asterisk-users] res_ODBC and failover (Bryant Zimmerman)

2011-09-28 Thread Reuben Fine 
  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 +
From: salaheddine elharit 
Subject: [asterisk-users] Limit outbond calls duration to 1 minute
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID:

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 +
From: Reuben Fine 
Subject: [asterisk-users] FreeTDS and MS-SQL with Asterisk RealTime
To: "asterisk-users@lists.digium.com"

Message-ID:

<8bb5d54201ff7a45ab99a14d0c37c166077...@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 
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID: <4e83611c.1010...@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 +
From: Tarek Sawah 
Subject: Re: [asterisk-users] Limit outbond calls duration to 1 minute
To: Asterisk Users 
Message-ID: 
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 +
From: salah.elharit...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Limit outbond calls duration to 1 minute

hello

Re: [asterisk-users] Anybody using BinFone Telecom?

2011-09-28 Thread John Novack 

I use Voip.ms and have a friend who uses BinFone. We both use IAX

He has had some issues lately, but it is unclear if it is binfone or his ISP. 
Losing internet connection and BinFone seems to fail to reconnect when his 
connection returns.

I have had no complaints with voip.ms
they have an excellent website with many options and easy configuration


John Novack



ft...@mindspring.com wrote:

Does anyone have any experience with BinFone for IAX termination?

They good look on the website, but I'm looking for any comments.

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[asterisk-users] I can't figure out how to redirect a call to a trunk.

2011-09-28 Thread Tomoki Taniguchi 
OK, i am hoping that someone will be able to help me out.
I am using FreePBX 2.8.1.4

I have two asterisk servers connected with a iax trunk.
The trunk is working fine when used via the outbound route setting.
meaning an extension on one server can call a specific extension on
the other server.

now what i want to do is set it up so that an incoming call (from a
third server)
is redirected to the second server from the first server when the
extension number does exist on the first server.
I thought I knew how to do this setup an inbound route that
answers any inbouond call and set the desitnation
to be "trunk" "server2"

but this doesn't seem to be working.
I know the inbound route is being called, i can even see the first
server trying to forward the call to server 2.
but the problem seems to be that no $OUTNUM is being set, so no
extension number on the second server is specified...
at which point the call fails.  I get an "all circuits are busy now"
announcement.

i upgraded to freepbx 2.8.x.x because of the fact that "FOLLOWME" and
"INBOUND ROUTES" were both able to specify a trunk
as a desitnation... but neither are working for me now?

am i doing something wrong?

-- 
Tomoki Taniguchi

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Re: [asterisk-users] PSTN connectivity

2011-09-28 Thread michael k 
Hi,

  Please see the sample.

A ) Analog HardwareType Ports Action   FXO Ports 1
Edit
 FXS
Ports --

B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: * from-analog*

*
C ) ZAP Trunk (DAHDI compatibility Mode)*


Trunk Description:
Outbound Caller ID:CID Options:
  Maximum Channels:   Disable Trunk:  Disable  Monitor Trunk Failures:
Enable   Outgoing Dial Rules   Dial Rules: 0471+NXX
  Dial Rules Wizards:
  Outbound Dial Prefix:Outgoing Settings   Zap Identifier (trunk name):


*D ) INBOUND route *

 Description:
Extensions: 199
*

E ) **OUTBOUND Route*

Route Name:  9_outside  Route CID:  Override Extension CID  Route
Password:  PIN
Set:
 Emergency Dialing:  Intra Company Route:  Music On Hold?
  Dial Patterns
8|NXXNXX 8|NXX
  Dial patterns wizards*: *
  Trunk SequenceZAP/g0  0
*
F ) In command Line I can see the following things *


[root@astrisks ~]# *dahdi_cfg -vv*


DAHDI Tools Version - 2.3.0

DAHDI Version: 2.3.0.1
Echo Canceller(s):
Configuration
==


Channel map:

Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01)

1 channels to configure.

Setting echocan for channel 1 to none


[root@astrisks ~]# *dahdi_scan*

[1]
active=yes
alarms=OK
description=Wildcard X100P Board 1
name=WCFXO/0
manufacturer=Digium
devicetype=Wildcard X100P
location=PCI Bus 02 Slot 02
basechan=1
totchans=1
irq=193
type=analog
port=1,FXO



*Asterisk CLI*


*astrisks*CLI> dahdi show status*

Description  Alarms  IRQbpviol CRC4   Fra
Codi Options  LBO
Wildcard X100P Board 1   OK  0  0  0  CAS
Unk   0 db (CSU)/0-133 feet (DSX-1)

*
output when i dialing to a local number*

Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890)
Verbosity is at least 3
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
-- Executing [s@from-internal:1] Macro("SIP/199-003a", "hangupcall")
in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/199-003a",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/199-003a",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/199-003a",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/199-003a", "") in
new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/199-003a' in macro 'hangupcall'
  == Spawn extension (from-internal, s, 1) exited non-zero on
'SIP/199-003a'
-- Executing [h@from-internal:1] Macro("SIP/199-003a", "hangupcall")
in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/199-003a",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/199-003a",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/199-003a",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
   -- Executing [s@macro-hangupcall:9] Hangup("SIP/199-003a", "") in new
stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/199-003a' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/199-003a'
















On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind  wrote:

> Some CLI logs will get you better help on the issue ! also paste the FXO
> configurations and how you configured it !
>
> On Wed, Sep 28, 2011 at 2:11 PM, michael k  wrote:
>
>> Hi All,
>>
>>   I am trying to connect my asterisk box with freepbx to PSTN. I
>> have purchased x100p FXO card and installed in my asterisk server. My
>> freepbx detected the x100p FXO card and i can see the card specific details
>> in command line. I have configured the following things.
>>
>> 1. OUTBOUND caller id and Dialing rules in Freepbx.
>>
>> 2. INBOUND route
>>
>> When i call to the PSTN number before connecting to the FXO card, i am
>> getting a ringing. But i get a message like the "number is out of order"
>> when i just connect the line to FXO card.
>>
>> Please some one help me to resolve his issue
>>
>> --
>> _
>> -- 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:
>

Re: [asterisk-users] PSTN connectivity

2011-09-28 Thread Sam Govind 
The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI.
there is some misconfiguration in FreePBX and your dialled number is not
hitting any dial-able rule.  See your FreePBX guide.


On Thu, Sep 29, 2011 at 11:01 AM, michael k  wrote:

> Hi,
>
>   Please see the sample.
>
> A ) Analog HardwareType Ports Action   FXO Ports 1 
> Edit
>   FXS
> Ports --
>
> B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: * from-analog*
>
> *
> C ) ZAP Trunk (DAHDI compatibility Mode)*
>
>
> Trunk Description:
> Outbound Caller ID:CID Options:
>   Maximum Channels:   Disable Trunk:  Disable  Monitor Trunk Failures:
> Enable   Outgoing Dial Rules   Dial Rules: 0471+NXX
>   Dial Rules Wizards:
>   Outbound Dial Prefix:Outgoing Settings   Zap Identifier (trunk name):
>
>
>
> *D ) INBOUND route *
>
>  Description:
> Extensions: 199
> *
>
> E ) **OUTBOUND Route*
>
> Route Name:  9_outside  Route CID:  Override Extension CID  Route
> Password:  PIN Set:
>  Emergency Dialing:  Intra Company Route:  Music On Hold?
>   Dial Patterns
> 8|NXXNXX 8|NXX
>   Dial patterns wizards*: *
>   Trunk SequenceZAP/g0  0
> *
> F ) In command Line I can see the following things *
>
>
> [root@astrisks ~]# *dahdi_cfg -vv*
>
>
> DAHDI Tools Version - 2.3.0
>
> DAHDI Version: 2.3.0.1
> Echo Canceller(s):
> Configuration
> ==
>
>
> Channel map:
>
> Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01)
>
> 1 channels to configure.
>
> Setting echocan for channel 1 to none
>
>
> [root@astrisks ~]# *dahdi_scan*
>
> [1]
> active=yes
> alarms=OK
> description=Wildcard X100P Board 1
> name=WCFXO/0
> manufacturer=Digium
> devicetype=Wildcard X100P
> location=PCI Bus 02 Slot 02
> basechan=1
> totchans=1
> irq=193
> type=analog
> port=1,FXO
>
>
>
> *Asterisk CLI*
>
>
> *astrisks*CLI> dahdi show status*
>
> Description  Alarms  IRQbpviol CRC4   Fra
> Codi Options  LBO
> Wildcard X100P Board 1   OK  0  0  0  CAS
> Unk   0 db (CSU)/0-133 feet (DSX-1)
>
> *
> output when i dialing to a local number*
>
> Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890)
> Verbosity is at least 3
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
> -- Executing [s@from-internal:1] Macro("SIP/199-003a",
> "hangupcall") in new stack
> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/199-003a",
> "1?skiprg") in new stack
> -- Goto (macro-hangupcall,s,4)
> -- Executing [s@macro-hangupcall:4] GotoIf("SIP/199-003a",
> "1?skipblkvm") in new stack
> -- Goto (macro-hangupcall,s,7)
> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/199-003a",
> "1?theend") in new stack
> -- Goto (macro-hangupcall,s,9)
> -- Executing [s@macro-hangupcall:9] Hangup("SIP/199-003a", "") in
> new stack
>   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
> 'SIP/199-003a' in macro 'hangupcall'
>   == Spawn extension (from-internal, s, 1) exited non-zero on
> 'SIP/199-003a'
> -- Executing [h@from-internal:1] Macro("SIP/199-003a",
> "hangupcall") in new stack
> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/199-003a",
> "1?skiprg") in new stack
> -- Goto (macro-hangupcall,s,4)
> -- Executing [s@macro-hangupcall:4] GotoIf("SIP/199-003a",
> "1?skipblkvm") in new stack
> -- Goto (macro-hangupcall,s,7)
> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/199-003a",
> "1?theend") in new stack
> -- Goto (macro-hangupcall,s,9)
>-- Executing [s@macro-hangupcall:9] Hangup("SIP/199-003a", "") in
> new stack
>   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
> 'SIP/199-003a' in macro 'hangupcall'
>   == Spawn extension (from-internal, h, 1) exited non-zero on
> 'SIP/199-003a'
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind  wrote:
>
>> Some CLI logs will get you better help on the issue ! also paste the FXO
>> configurations and how you configured it !
>>
>> On Wed, Sep 28, 2011 at 2:11 PM, michael k  wrote:
>>
>>> Hi All,
>>>
>>>   I am trying to connect my asterisk box with freepbx to PSTN. I
>>> have purchased x100p FXO card and installed in my asterisk server. My
>>> freepbx detected the x100p FXO card and i can see the card specific details
>>> in command line. I have configured the following things.
>>>
>>> 1. OUTBOUND caller id and Dialing rules in Freepbx.
>>>
>>> 2. INBOUND route
>>>
>>> When i call to the PSTN number before connecting to the FXO card, i am
>>> getting a ringing. But i get a message like the "number is out of order"
>>> when i just connect the line to FXO card.
>>>
>>> Please some one help me to resolve his issue
>>>
>>> --
>>> _
>>> -- Bandwidth and Coloca

Re: [asterisk-users] PSTN connectivity

2011-09-28 Thread michael k 
Can you please figure out the configuration issue in my freepbx ?




On Thu, Sep 29, 2011 at 11:35 AM, Sam Govind  wrote:

> The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI.
> there is some misconfiguration in FreePBX and your dialled number is not
> hitting any dial-able rule.  See your FreePBX guide.
>
>
> On Thu, Sep 29, 2011 at 11:01 AM, michael k  wrote:
>
>> Hi,
>>
>>   Please see the sample.
>>
>> A ) Analog HardwareType Ports Action   FXO Ports 1 
>> Edit
>>   FXS
>> Ports --
>>
>> B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: * from-analog
>> *
>>
>> *
>> C ) ZAP Trunk (DAHDI compatibility Mode)*
>>
>>
>> Trunk Description:
>> Outbound Caller ID:CID Options:
>>   Maximum Channels:   Disable Trunk:  Disable  Monitor Trunk Failures:
>> Enable   Outgoing Dial Rules   Dial Rules: 0471+NXX
>>   Dial Rules Wizards:
>>   Outbound Dial Prefix:Outgoing Settings   Zap Identifier (trunk
>> name):
>>
>>
>> *D ) INBOUND route *
>>
>>  Description:
>> Extensions: 199
>> *
>>
>> E ) **OUTBOUND Route*
>>
>> Route Name:  9_outside  Route CID:  Override Extension CID  Route
>> Password:  PIN Set:
>>  Emergency Dialing:  Intra Company Route:  Music On Hold?
>>   Dial Patterns
>> 8|NXXNXX 8|NXX
>>   Dial patterns wizards*: *
>>   Trunk SequenceZAP/g0  0
>> *
>> F ) In command Line I can see the following things *
>>
>>
>> [root@astrisks ~]# *dahdi_cfg -vv*
>>
>>
>> DAHDI Tools Version - 2.3.0
>>
>> DAHDI Version: 2.3.0.1
>> Echo Canceller(s):
>> Configuration
>> ==
>>
>>
>> Channel map:
>>
>> Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01)
>>
>> 1 channels to configure.
>>
>> Setting echocan for channel 1 to none
>>
>>
>> [root@astrisks ~]# *dahdi_scan*
>>
>> [1]
>> active=yes
>> alarms=OK
>> description=Wildcard X100P Board 1
>> name=WCFXO/0
>> manufacturer=Digium
>> devicetype=Wildcard X100P
>> location=PCI Bus 02 Slot 02
>> basechan=1
>> totchans=1
>> irq=193
>> type=analog
>> port=1,FXO
>>
>>
>>
>> *Asterisk CLI*
>>
>>
>> *astrisks*CLI> dahdi show status*
>>
>> Description  Alarms  IRQbpviol CRC4   Fra
>> Codi Options  LBO
>> Wildcard X100P Board 1   OK  0  0  0  CAS
>> Unk   0 db (CSU)/0-133 feet (DSX-1)
>>
>> *
>> output when i dialing to a local number*
>>
>> Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890)
>> Verbosity is at least 3
>>   == Using SIP RTP TOS bits 184
>>   == Using SIP RTP CoS mark 5
>> -- Executing [s@from-internal:1] Macro("SIP/199-003a",
>> "hangupcall") in new stack
>> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/199-003a",
>> "1?skiprg") in new stack
>> -- Goto (macro-hangupcall,s,4)
>> -- Executing [s@macro-hangupcall:4] GotoIf("SIP/199-003a",
>> "1?skipblkvm") in new stack
>> -- Goto (macro-hangupcall,s,7)
>> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/199-003a",
>> "1?theend") in new stack
>> -- Goto (macro-hangupcall,s,9)
>> -- Executing [s@macro-hangupcall:9] Hangup("SIP/199-003a", "") in
>> new stack
>>   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
>> 'SIP/199-003a' in macro 'hangupcall'
>>   == Spawn extension (from-internal, s, 1) exited non-zero on
>> 'SIP/199-003a'
>> -- Executing [h@from-internal:1] Macro("SIP/199-003a",
>> "hangupcall") in new stack
>> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/199-003a",
>> "1?skiprg") in new stack
>> -- Goto (macro-hangupcall,s,4)
>> -- Executing [s@macro-hangupcall:4] GotoIf("SIP/199-003a",
>> "1?skipblkvm") in new stack
>> -- Goto (macro-hangupcall,s,7)
>> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/199-003a",
>> "1?theend") in new stack
>> -- Goto (macro-hangupcall,s,9)
>>-- Executing [s@macro-hangupcall:9] Hangup("SIP/199-003a", "") in
>> new stack
>>   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
>> 'SIP/199-003a' in macro 'hangupcall'
>>   == Spawn extension (from-internal, h, 1) exited non-zero on
>> 'SIP/199-003a'
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind  wrote:
>>
>>> Some CLI logs will get you better help on the issue ! also paste the FXO
>>> configurations and how you configured it !
>>>
>>> On Wed, Sep 28, 2011 at 2:11 PM, michael k  wrote:
>>>
 Hi All,

   I am trying to connect my asterisk box with freepbx to PSTN. I
 have purchased x100p FXO card and installed in my asterisk server. My
 freepbx detected the x100p FXO card and i can see the card specific details
 in command line. I have configured the following things.

 1. OUTBOUND caller id and Dialing rules in Freepbx.

 2. INBOUND route

 When i call to the PSTN number before connecting to