Re: [asterisk-users] Find out what context is the exten from

[Tiago Geada]

Hi

​A extension can existe in more than one context like special extensions
(for instance s or i or h)​

anyway you can execute "dialplan show context" in the asterisk cli
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Re: [asterisk-users] Replacement for phpagi?

[Tiago Geada]

Hi
I would recommend PAMI - its object oriented and well structured

On 10 August 2016 at 19:49, Alex Villací­s Lasso 
wrote:

> El 10/08/16 a las 12:06, Carlos Chavez escribió:
>
>> Anyone know a good replacement for phpagi? Unfortunately development
>> stalled long ago and it has not been updated.  What is the best solution
>> for AMI and AGI on PHP? Thanks.
>>
>>
>> In the case of AMI, you could use the AMI client from the Elastix
> CallCenter dialer daemon:
>
> https://sourceforge.net/p/elastix/code/HEAD/tree/branches/2.
> 5.0/apps/extras/callcenter/setup/dialer_process/dialer/
> AMIClientConn.class.php
>
> This class was once based on phpagi-asmanager.php but has since been
> completely rewritten to make use of an internal non-blocking I/O model. The
> main internal client is the AMIEventProcess class in the same project
> directory.
>
>
>
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Re: [asterisk-users] Queues - periodic announce while ringing members

[Tiago Geada]

Hi, what I did, I mixed the music on hold to have the announce in at a
specific time without leaving queue

On 25 February 2016 at 16:53, Daniel Chavez  wrote:

> Ish,
> I use the same version of Asterisk on CentOS 6.7. I wonder the same thing.
> Hopefully we will find this out.
>
>
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Re: [asterisk-users] Live Recording on the NAS?

[Tiago Geada]

Hi,

MixMonitor(filename.extension[,options[,command]])


you can run a shell command that moves the file to its final location,
after being written on ramdisk. This seems the simple way to do it

On 9 October 2015 at 11:53, jg  wrote:

>
> I am planning to move Asterisk from physical server to a VM on a ESXi
> host.
>
> VMware datastore / VM's will be stored on the shared storage on the NAS
> (NSF). I might get Synology NAS.
>
> Do you store call live recording on the NAS? There would be around 60
> concurrent calls recording at the same time and it may cause network
> bottleneck.
>
> There will be other VM's stored on the NAS like Windows Servers, Linux
> Servers, Database, etc.
>
> 60 concurrent alls sounds like a lot. I'd work with a RAM-disk and some
> post-processing to be safe. I have a low priority background task that
> moves finished sound files to a file server and converts them to mp3. The
> software that accesses the audio looks for both formats at both places. I
> think it is generally a good idea to handle file issues outside of Asterisk.
>
> jg
>
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Re: [asterisk-users] Anyone doing speech to text?

[Tiago Geada]

​You're right, I misinterpreted

Sorry!​
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Re: [asterisk-users] Anyone doing speech to text?

[Tiago Geada]

I had been using google tts, but it started requiring a captcha for my
browser, and via linux I can't access
http://translate.google.com/translate_tts?q=test (redirects to captcha)

as so, its not reliable

On 27 August 2015 at 17:16, Carlos Chavez cur...@telecomabmex.com wrote:

 On 8/26/15 1:15 PM, Tech Support wrote:

 All;

I have a customer who is looking for a good speech to text solution,
 either open source or reasonably priced commercial product, I’m open to
 suggestions.

 Thanks;

 John V



 For a commercial option try Lumenvox, had very good results.  For free
 you can try google tts but you never know when google will decide to pull
 the plug on something.

 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52
 (55)9116-91161


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Re: [asterisk-users] asterisk email to fax

[Tiago Geada]

we use a PHP web page, that takes a few formats, PDF being the most common,
anc convert it to TIFF.

If conversion succeeds we allow to download the TIFF file as a preview.
Then the user confirms and the PHP places a .call file in asterisk spool

On 25 June 2015 at 19:51, Ryan, Travis ry...@oscarwinski.com wrote:

  I hope his mother in law doesn’t live with him. That’s a support issue
 for sure.





 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kevin Larsen
 *Sent:* Thursday, June 25, 2015 2:50 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] asterisk email to fax



  Since the O.P. said he's using it for his home office, I think he'll
  be able to control user expectations :-)


 I provide tech support to my parents on all their computers. The amount of
 annoyance I have dealt with in the last few months over the fact that a
 recipe program and various card making programs designed for Windows 3.1/95
 won't run on my mom's Windows 7 64 bit computer tells me you are not as
 right as you think you are.

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Re: [asterisk-users] Preserve CDR unique across multiple servers

[Tiago Geada]

Hi,

We use sip headers to send the linkedid across servers, and place it into
CDR as remoteLinkedId

On 26 June 2015 at 15:18, Rui Mota ruim...@gmail.com wrote:

 I am already using the unique in both servers, but both generate different
 id’s, but i am unable to get the original one from the gateway box to store
 it in the final CDR…

 --
 Rui Mota
 Sent with Sparrow http://www.sparrowmailapp.com/?sig

 On Friday 26 June 2015 at 14:52, Tech Support wrote:

 Check out the “uniqueid” parameter in cdr.conf and cdr_adaptive_odbc.conf.

 John V.



 *From:* asterisk-users-boun...@lists.digium.com [
 mailto:asterisk-users-boun...@lists.digium.com
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rui Mota
 *Sent:* Friday, June 26, 2015 7:05 AM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] Preserve CDR unique across multiple servers



 Hi.



 I am using two servers in my configuration: one for phones registration
 and another one as gateway, where all the providers are connected. Both are
 connected through an IAX trunk.

 I am having some trouble on matching both CDR’s, since durations for a
 call are not always the same in both servers, start/end date time are
 sometimes also different, etc.



 Is there any way to send the uniqueid of the original call, maybe through
 the IAX trunk, and get it on the gateway server to save it in the final
 CDR’s userfield?  That way they would match by that field.



 Thank you in advance.



 --

 Rui Mota


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Re: [asterisk-users] Asterisk 13 logging to two places

[Tiago Geada]

messages = error

states to log error messages to 'messages' log file


On 26 June 2015 at 17:50, Tom Peters tpet...@mcts.org wrote:

 Switched from Asterisk 1.8 to 13.3.2. Now it logs to
 /var/log/asterisk/full (good) as well as /var/log/messages (not good).
 Anyone know why?

 # grep -v ^; logger.conf

 [general]
 [logfiles]
 console = notice,warning,error
 messages = error
 full = notice,warning,error,debug,verbose,dtmf,fax

 Thankfully, the .../full logs are rotating properly now (thanks Dale) but
 we don't need both files cluttered up. We use /var/log/asterisk/full pretty
 extensively for troubleshooting, but I want /var/log/messages for other
 stuff. Didn't do this under the old version.

 Any other files you want to see?

 Running on CentOS release 7.1.1503



 Thomas M. Peters | Systems Administrator | tpet...@mcts.org
 Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org



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Re: [asterisk-users] Re-INVITE and bridge breakage

[Tiago Geada]

If I'm not mistaken, canreinvite=no is now directmedia=no 
But check other values of directmedia 

Sent from my iPhone

 On 15 May 2015, at 19:21, Luca Pradovera luca.pradov...@gmail.com wrote:
 
 Hello,
 as a variation of our issues with Adhearsion calls dropping when an INVITE 
 comes in for a bridged call, I now have a new issue to contend with.
 
 Our call is in an AsyncAGI application, and has been bridged to another 
 channel.
 The provider that supplies the DID sends a polling reINVITE every 15 minutes 
 (it's a documented Metaswitch behavior amongst others).
 The reINVITE is seen as a new offer by Adhearsion, which then drops the call 
 on trying to re-bridge the two channels.
 
 Is there any way to specify that reINVITEs are not to be accepted at the 
 Asterisk level?
 
 Thanks,
 
 Luca
 
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Re: [asterisk-users] Custom UUID in originate and AMI

[Tiago Geada]

what do you mean by set

you can use like:

Variable: __CUSTOMID=UUID-string\r\n

to be able to read back ${CUSTOMID} back in the dialplan ... ?

On 8 May 2015 at 19:04, Mehdi Shirazi mahdi_shir...@yahoo.com wrote:

 Hi
 Could someone please help me how to set Custom generated UUID in Originate
 action in AMI ?

 Regards
 Babak


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Re: [asterisk-users] Checking for human answer

[Tiago Geada]

Hello


We use originate that places a call in a queue (channel parameter is a
Local/dialplan)

When the call is answered in queue, it is bridged with the operator, and
then starts the second channel leg: Dial out to wherever trough local
channel


we set a sip header with dialstatus, so if the operator hangs the call, we
see a CANCEL back in our pbx


On 20 July 2014 17:20, Valter Nogueira vgnogue...@gmail.com wrote:

 In fact, Asterisk console shows a message warning that call is not
 finished because of the macro leg




 2014-07-20 13:19 GMT-03:00 Valter Nogueira vgnogue...@gmail.com:

 No, I am testing with IP phones.

 When caller hangs-out the macro is not aware - but when calle hangs the
 macro is.


 2014-07-20 12:31 GMT-03:00 Doug Lytle supp...@drdos.info:

 Valter Nogueira wrote:

 The problem is in the opposite side - when someone call us and hangs
 before the operator press the number.


 Then my guess would be you're on analog lines?

 Without call supervision on the line, there will be no way of detecting
 when an analog call has been dropped, other then when the operator has
 decided there is nobody there and hangs up at which point the call should
 be dropped.

 Digital lines and VOIP lines shouldn't have this issue since they have
 call supervision.


 Doug




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Re: [asterisk-users] CDR custom variable on second call leg - via originate or .call file

[Tiago Geada]

Hi Richard.

I looked at both pages, yes.


My goal is to have a flag on the cdr database records when the call is not
yet connected to the second leg.

So, the Channel argument takes a call to a portion of dialplan that will
try several steps. And at those steps the custom variable will be set to
'foo'

When answered, the Context and Extension argument take the call trough
another piece of dialplan that will have that its CDR entries with the
custom variable set to 'bar'

Just like on the example stated!


However, only 'foo' gets written!


On 27 June 2014 21:16, Richard Mudgett rmudg...@digium.com wrote:




 On Fri, Jun 27, 2014 at 1:30 PM, Tiago Geada tiago.ge...@gmail.com
 wrote:

 Is there something I can do regarding this issue?


 Have you looked at these wiki pages?
 https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files
 The setvar parameter may help here.

 https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance

 Richard


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Re: [asterisk-users] Need to spoof the callerid using the AMI Originate

[Tiago Geada]

Hi,

You can use a Local channel in your  originate, and have a piece of local
dialplan change that for you. Set(CALLERID(num)=x)


On 13 June 2014 15:32, Dan Cropp d...@amtelco.com wrote:

 We have several customers we need to place outbound calls for (in a single
 system).  May have to place calls for thousands of different caller ids.
 Customer signs a contract guaranteeing the caller id they provide is owned
 by them.



 I have everything setup for AMI Originate and can place the calls.



 However, I’m encountering a problem with the caller id.

 The system I’m dialing through requires my contact to be something like
 1234@xyz.



 If I set the CallerID to something like Jane Doe 1234 it will correctly
 set my Contact so the system accepts the call and it dials the number.

 However, the SIP INVITE message From field is set to “Jane Doe”
 1...@xxx.xxx.xxx.xxx

 Is there a way to make the SIP INVITE message have different caller id
 values for the From and the Contact fields?



 Additionally, is it possible to set the callerid number value to a PSTN
 number instead of a SIP number@domain?



 I tried setting the callerid(num) via the variable field, but that doesn’t
 seem to work.

 Below is a sample of the AMI Originate message I’m sending.



 Action: Originate

 ActionID: MyAction

 Channel: SIP/xxx.xxx.xxx.xxx/1234567890

 Exten: testing

 Context: MyContext

 Priority: 1

 Timeout: 3

 CallerID: Jane Doe 123

 Variable: CALLERID(num)=222333

 Async: true





 Have a great day!

 Dan

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Re: [asterisk-users] Need to spoof the callerid using the AMI Originate

[Tiago Geada]

Actually you shold do that on MyContext


On 6 July 2014 19:21, Tiago Geada tiago.ge...@gmail.com wrote:

 Hi,

 You can use a Local channel in your  originate, and have a piece of local
 dialplan change that for you. Set(CALLERID(num)=x)


 On 13 June 2014 15:32, Dan Cropp d...@amtelco.com wrote:

 We have several customers we need to place outbound calls for (in a
 single system).  May have to place calls for thousands of different caller
 ids.  Customer signs a contract guaranteeing the caller id they provide is
 owned by them.



 I have everything setup for AMI Originate and can place the calls.



 However, I’m encountering a problem with the caller id.

 The system I’m dialing through requires my contact to be something like
 1234@xyz.



 If I set the CallerID to something like Jane Doe 1234 it will correctly
 set my Contact so the system accepts the call and it dials the number.

 However, the SIP INVITE message From field is set to “Jane Doe”
 1...@xxx.xxx.xxx.xxx

 Is there a way to make the SIP INVITE message have different caller id
 values for the From and the Contact fields?



 Additionally, is it possible to set the callerid number value to a PSTN
 number instead of a SIP number@domain?



 I tried setting the callerid(num) via the variable field, but that
 doesn’t seem to work.

 Below is a sample of the AMI Originate message I’m sending.



 Action: Originate

 ActionID: MyAction

 Channel: SIP/xxx.xxx.xxx.xxx/1234567890

 Exten: testing

 Context: MyContext

 Priority: 1

 Timeout: 3

 CallerID: Jane Doe 123

 Variable: CALLERID(num)=222333

 Async: true





 Have a great day!

 Dan

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Re: [asterisk-users] recording in mp3

[Tiago Geada]

no need.


mixmonitor has a argument that is a script ran just as the recording is
finished.

we use this to move the file from ramfs to final destination.

you can use it to use sox and convert it...


On 2 July 2014 18:54, Dave Platt dpl...@radagast.org wrote:


  Problem with this is client needs to listen to the call recordings and
 my interface will only display .wav or .mp3 so they will moan if they have
 to wait until the next day for today's recordings

 If you're up to writing a bit of shell script, and are running
 on Linux, you could automate the conversion process so that it
 happens as soon as the recording is completed.

 Look at the inotify system service (man section 7) and the
 inotifywatch program.  You can tell inotifywatch to monitor
 files being written into a specific directory (or set of
 directories) and output a series of events when files in this
 directory are open or closed.

 What you'd probably want to do, is catch the close_write
 events (a file has been closed, and it had been opened in
 a mode which allows it to be written).  When you see a
 close_write event for a recording file of the sort that
 Asterisk writes, you'd check to see if it's been converted
 to your desired format yet.  If not, fire off a separate
 task (e.g. via batch) to convert it.

 Here's a very simple script I did to do something like this...
 run a periodic-processing script a few seconds after files
 with a specific name pattern have been touched in any way.
 It's not sophisticated enough to look only for close or
 close_wait events, but it should give you the idea.

 #!/bin/bash

 function processevents () {
  action=0
  while true ; do
if [ $action == 0 ] ; then
timeout=300
else
timeout=5
fi
read -t $timeout event
if [ $? != 0 ] ; then
   action=0
   /data/soundchaser/periodic
else
   if [[ $event =~ .wav || $event =~ .gotit ]] ; then
   action=1
   fi
fi
  done
 }

 cd /data/soundchaser

 inotifywait -m /data/soundchaser/public_html/done | processevents


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Re: [asterisk-users] CDR custom variable on second call leg - via originate or .call file

[Tiago Geada]

Is there something I can do regarding this issue?


On 16 June 2014 11:39, Tiago Geada tiago.ge...@gmail.com wrote:

 Hi,

 Thank you for your explanation about channel halds .. These .call files
 are always different from other calls.

 Well I would like some custom var to have a piece of information while it
 is queuing, and another piece of information, once answered in queue, thus
 just before dialing to context outbound.

 the outbound cdr bit, is fine. I'm now interested in the -
 Context,Extension - or the ;1 half of the channel. Here I would like to set
 remoteUid=bar but although the Set() is there and shown in verbosity, the
 insert query doesn't take it in.

 The CDR bit with remoteUid=foo is OK, the bit that should have
 remoteUid=bar is not




 On 11 June 2014 19:24, Matthew Jordan mjor...@digium.com wrote:




 On Wed, Jun 11, 2014 at 9:10 AM, Tiago Geada tiago.ge...@gmail.com
 wrote:

 Hi,

 Let me append some extra info

 cdr variable foo, shows on database, but value 'bar' doens't

 its not even shown in the insert query

 I tried with master_channel but no change


 I think you need to be a bit more specific about what CDR records you're
 getting and what you'd like to have happen.

 You have the following call file:

 snip






 ## test call file



 Channel: Local/queue@TiagoGeada

 CallerID: teste-geada:0:210332450:

 MaxRetries: 0

 RetryTime: 1

 WaitTime: 8640

 Account: teste-geada

 Context: TiagoGeada

 Extension: outbound

 Archive: Yes






 This will create a Local channel with two halves. The ;2 half will
 execute in the dialplan at TiagoGeada,queue,1 - the ;1 half will execute in
 the dialplan at TiagoGeada,outbound,1. The ;2 Local channel will execute
 first until it is Answered; once Answered, that will trigger the ;1 half to
 start execution. That will create two CDRs, one for each Local channel half.

 MASTER_CHANNEL won't apply here, as MASTER_CHANNEL only applies to a
 Parent/Child relationship between channels, that is, when one channel has
 created another channel. This occurs when a channel dials another channel.
 The ;1 side didn't create the ;2 side, they are effectively two sides of
 the same channel.







 ## dialplan



 queue = {

 Set(CDR(remoteUid)=foo);

 Queue(TiagoGeada,t,,,100);

 Hangup();

 }



 outbound = {

 //NoCDR();

 //ForkCDR(vdD);

 //ResetCDR(v);

 Set(CDR(remoteUid,r)=bar);

 Dial(Local/932485457@outbound,,gT);

 Hangup();

 }



 Looking at your Dialplan for the outbound extension, you dial yet another
 Local channel. I would expect this to result in 3 CDR entries:

 Source Channel Destination Channel
 Local/queue@TiagoGeada;2
  Local/queue@TiagoGeada;1   Local/932485427@outbound;1
 Local/932485457@outbound;2

 So, the question is, which CDR are you talking about? What value do you
 want where? Keep in mind that unless all channels are answered, they won't
 show up in your CDRs (unless you have unanswered=yes set in cdr.conf).

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

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Re: [asterisk-users] CDR custom variable on second call leg - via originate or .call file

[Tiago Geada]

Hi,

Thank you for your explanation about channel halds .. These .call files are
always different from other calls.

Well I would like some custom var to have a piece of information while it
is queuing, and another piece of information, once answered in queue, thus
just before dialing to context outbound.

the outbound cdr bit, is fine. I'm now interested in the -
Context,Extension - or the ;1 half of the channel. Here I would like to set
remoteUid=bar but although the Set() is there and shown in verbosity, the
insert query doesn't take it in.

The CDR bit with remoteUid=foo is OK, the bit that should have
remoteUid=bar is not




On 11 June 2014 19:24, Matthew Jordan mjor...@digium.com wrote:




 On Wed, Jun 11, 2014 at 9:10 AM, Tiago Geada tiago.ge...@gmail.com
 wrote:

 Hi,

 Let me append some extra info

 cdr variable foo, shows on database, but value 'bar' doens't

 its not even shown in the insert query

 I tried with master_channel but no change


 I think you need to be a bit more specific about what CDR records you're
 getting and what you'd like to have happen.

 You have the following call file:

 snip






 ## test call file



 Channel: Local/queue@TiagoGeada

 CallerID: teste-geada:0:210332450:

 MaxRetries: 0

 RetryTime: 1

 WaitTime: 8640

 Account: teste-geada

 Context: TiagoGeada

 Extension: outbound

 Archive: Yes






 This will create a Local channel with two halves. The ;2 half will execute
 in the dialplan at TiagoGeada,queue,1 - the ;1 half will execute in the
 dialplan at TiagoGeada,outbound,1. The ;2 Local channel will execute first
 until it is Answered; once Answered, that will trigger the ;1 half to start
 execution. That will create two CDRs, one for each Local channel half.

 MASTER_CHANNEL won't apply here, as MASTER_CHANNEL only applies to a
 Parent/Child relationship between channels, that is, when one channel has
 created another channel. This occurs when a channel dials another channel.
 The ;1 side didn't create the ;2 side, they are effectively two sides of
 the same channel.







 ## dialplan



 queue = {

 Set(CDR(remoteUid)=foo);

 Queue(TiagoGeada,t,,,100);

 Hangup();

 }



 outbound = {

 //NoCDR();

 //ForkCDR(vdD);

 //ResetCDR(v);

 Set(CDR(remoteUid,r)=bar);

 Dial(Local/932485457@outbound,,gT);

 Hangup();

 }



 Looking at your Dialplan for the outbound extension, you dial yet another
 Local channel. I would expect this to result in 3 CDR entries:

 Source Channel Destination Channel
 Local/queue@TiagoGeada;2
 Local/queue@TiagoGeada;1   Local/932485427@outbound;1
 Local/932485457@outbound;2

 So, the question is, which CDR are you talking about? What value do you
 want where? Keep in mind that unless all channels are answered, they won't
 show up in your CDRs (unless you have unanswered=yes set in cdr.conf).

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

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Re: [asterisk-users] CDR custom variable on second call leg - via originate or .call file

[Tiago Geada]

Hi,

Let me append some extra info

cdr variable foo, shows on database, but value 'bar' doens't

its not even shown in the insert query

I tried with master_channel but no change


On 10 June 2014 16:25, Eric Wieling ewiel...@nyigc.com wrote:

 Using Set(MASTER_CHANNEL(CDR(remoteUid))=foo); might do what you want



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mikael Fredin
 *Sent:* Tuesday, June 10, 2014 11:18 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] CDR custom variable on second call leg -
 via originate or .call file



 As far as I know, only way to set variables on another channel would be:

  asterisk -rx core show help dialplan set chanvar
 Usage: dialplan set chanvar channel varname value
Set channel variable varname to value





 On 10 June 2014 16:39, Tiago Geada tiago.ge...@gmail.com wrote:

 Hi





 We have the following test .call file and test dialplan:



 I can't set a custom CDR var to a value on one channel leg, and another
 value on the connected channel leg?





 Is there a way I can woraround this issue?







 ## test call file



 Channel: Local/queue@TiagoGeada

 CallerID: teste-geada:0:210332450:

 MaxRetries: 0

 RetryTime: 1

 WaitTime: 8640

 Account: teste-geada

 Context: TiagoGeada

 Extension: outbound

 Archive: Yes









 ## dialplan



 queue = {

 Set(CDR(remoteUid)=foo);

 Queue(TiagoGeada,t,,,100);

 Hangup();

 }



 outbound = {

 //NoCDR();

 //ForkCDR(vdD);

 //ResetCDR(v);

 Set(CDR(remoteUid,r)=bar);

 Dial(Local/932485457@outbound,,gT);

 Hangup();

 }


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[asterisk-users] CDR custom variable on second call leg - via originate or .call file

[Tiago Geada]

Hi


We have the following test .call file and test dialplan:

I can't set a custom CDR var to a value on one channel leg, and another
value on the connected channel leg?


Is there a way I can woraround this issue?



## test call file

Channel: Local/queue@TiagoGeada
CallerID: teste-geada:0:210332450:
MaxRetries: 0
RetryTime: 1
WaitTime: 8640
Account: teste-geada
Context: TiagoGeada
Extension: outbound
Archive: Yes




## dialplan

queue = {
Set(CDR(remoteUid)=foo);
Queue(TiagoGeada,t,,,100);
Hangup();
}

outbound = {
//NoCDR();
//ForkCDR(vdD);
//ResetCDR(v);
Set(CDR(remoteUid,r)=bar);
Dial(Local/932485457@outbound,,gT);
Hangup();
}
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Re: [asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?

[Tiago Geada]

Hi all,


How does one detect the 'divert' to voicemail?

Say we have PRI lines and as wel as SIP Trunks to connect to mobile phones.
How can asterisk know if the call is being diverted??


On 14 February 2014 10:11, Chris Bagnall aster...@lists.minotaur.cc wrote:

 On 14/2/14 9:21 am, Gareth Blades wrote:

 I would suggest using the 'M' option on the Dial command to run a macro.
 The macro can just wait fir a key to be pressed and until it is pressed
 the Dial is still effectively ringing. So if it does go to voicemail
 then the call wont get put through. You need to make sure you have a
 suitable value set to abandon the agent call if its ringing too long.
 The callee may also find they are left multiple voicemail messages.


 This is the approach we've used in the past: force the recipient to hit a
 button to accept the call, something which their mobile voicemail will
 never be able to do.

 The alternative - and it only really applies if you have control of the
 mobiles in question - is to disable the mobile network's voicemail service
 entirely, and manage diverts from the handset. That way you can then
 recreate your own 'mobile voicemail' service on your asterisk platform with
 all the normal asterisk VM benefits such as email delivery, etc.

 You can then of course detect when those mobiles 'divert' to voicemail
 (since it's now on your system), and kick them out of the queue at that
 point.

 Kind regards,

 Chris
 --
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Re: [asterisk-users] IOPS required by Asterisk for Call Recording

[Tiago Geada]

Hi,

MixMonitor takes a parameter of a system command to run when the recording
finishes. Like Chris said, you can write to ramdisk, and run a script that
will move the file into final position only when the call has done recording

Here we use:
Set(recordFile=${UNIQUEID}_${NUMBER}.gsm);

Set(recordPath=/var/log/asterisk/recordings/${CALLERID(dnid)}/${STRFTIME(${EPOCH},GMT+0,%F)});

MixMonitor(/ramdrive/${recordFile},,/usr/local/bin/mixmon ${recordFile}
${recordPath});
SIPAddHeader(X-REC-FILE:
${recordPath}/${recordFile});

and /usr/local/bin/mixmon will move the file to $recordPath and whatever
else needs done on that file...



On 27 January 2014 21:55, Matthew Jordan mjor...@digium.com wrote:

 On Mon, Jan 27, 2014 at 1:02 PM, Ron Wheeler
 rwhee...@artifact-software.com wrote:
  Can you get a reading of the total number of I/Os during your test? Peak
  IOPS?
  That might tell you very quickly about the storage pattern that Asterisk
  uses.
 
  Can you configure a RAM drive to see if disk is really the bottleneck.
 May
  need to add some more RAM memory to your configuration.
 
  What is your network capacity? Usually one can write faster than the
 network
  can deliver - just to make sure that you are chasing the right
 bottleneck.
 
  What happens at 80 calls to tell you that you have run out of IOPS?

 Dovetailing on this question, I'll add one as well:

 Are you recording using MixMonitor, or Monitor?

 Depending on your answer to the what happens at 80 calls, you may
 get better results with MixMonitor over Monitor. MixMonitor offloads
 the recording of the media to a separate thread; Monitor attempts to
 record the audio on the thread servicing the channel(s).

 Matt

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

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[asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

Hi folks,

We've been having a weird issue... It is happening more often in the last
few months...

Most inbound calls, we have in our dialplan before Queue():

Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});

So when the call rings a member, softphone will show this string 

The issue is that sometimes the string showing in the softphone is not the
same. Its a string from a past call, in the latest case I've seen, from
about 40 days ago!!

User took a screenshot, I've searched for that uniqueid showing in
softphone in cdr, and that string was valid for a different call 40 days
ago!!


I searched full log, and Set() sets the correct string... I can't figure
why softphone shows a string from a past call !!

:(

Any hints ?
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

You're right, seems like a nice way to debug. Regarding that, how would the
impact be affected running it on asterisk box? I guess only port 5060 is
not too bad


On 16 January 2014 14:09, Gareth Blades mailinglist+aster...@dns99.co.ukwrote:

  On 16/01/14 10:47, Tiago Geada wrote:

  Hi folks,

  We've been having a weird issue... It is happening more often in the
 last few months...

  Most inbound calls, we have in our dialplan before Queue():

  Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});

  So when the call rings a member, softphone will show this string 

  The issue is that sometimes the string showing in the softphone is not
 the same. Its a string from a past call, in the latest case I've seen, from
 about 40 days ago!!

  User took a screenshot, I've searched for that uniqueid showing in
 softphone in cdr, and that string was valid for a different call 40 days
 ago!!


  I searched full log, and Set() sets the correct string... I can't figure
 why softphone shows a string from a past call !!

  :(

  Any hints ?


  I would leave tcpdump running capturing port 5060 so you can load it onto
 wireshark and have a look at the sip headers. That will tell you if the SIP
 is incorrect or if its a problem with the client.

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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

Second thought, that would only allow me to know if there is a problem on
asterisk or softphone.

Because the old callerid(name) that was presented on the softphone,
belonged to a call made to a different peer, I doubt that it would be a
softphone problem.

Our softphones are fixed with the same peer/extension .. if the wrong
callerid were originally called to the same peer.. I guess that would be
worth it..

even so, I will try and measure the impact on performance, however if
asterisk did send the wrong string, how could I debug that??


On 16 January 2014 14:27, Tiago Geada tiago.ge...@gmail.com wrote:

 You're right, seems like a nice way to debug. Regarding that, how would
 the impact be affected running it on asterisk box? I guess only port 5060
 is not too bad


 On 16 January 2014 14:09, Gareth Blades 
 mailinglist+aster...@dns99.co.ukwrote:

  On 16/01/14 10:47, Tiago Geada wrote:

  Hi folks,

  We've been having a weird issue... It is happening more often in the
 last few months...

  Most inbound calls, we have in our dialplan before Queue():

  Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL})
 ;

  So when the call rings a member, softphone will show this string 

  The issue is that sometimes the string showing in the softphone is not
 the same. Its a string from a past call, in the latest case I've seen, from
 about 40 days ago!!

  User took a screenshot, I've searched for that uniqueid showing in
 softphone in cdr, and that string was valid for a different call 40 days
 ago!!


  I searched full log, and Set() sets the correct string... I can't
 figure why softphone shows a string from a past call !!

  :(

  Any hints ?


  I would leave tcpdump running capturing port 5060 so you can load it
 onto wireshark and have a look at the sip headers. That will tell you if
 the SIP is incorrect or if its a problem with the client.

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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

Thank you Gareth

I will try that :)


On 16 January 2014 14:55, Gareth Blades mailinglist+aster...@dns99.co.ukwrote:

  Very little as the amount of data being captured is quite small. We have
 it running on our production servers which routinely handle a couple of
 hundred concurrent calls.

 This is the script we use to start off the capture. It uses rolling
 capture files so we will always have the last X number of capture logs. It
 works very well and we have a custom system which enables us to search for
 calls and request traces for them for when we have to diagnose problems.

 #!/bin/bash
 cd /var/lib/asterisk/siptraces
 DATE=`date +%Y%m%d%H%M%S`
 TRACEFILE=/var/lib/asterisk/siptraces/$DATE-siptrace.pcap
 nohup /usr/sbin/tcpdump -p -i eth0 -s 0 port 5060 -w $TRACEFILE -C 10 -W
 500 



 On 16/01/14 14:27, Tiago Geada wrote:

  You're right, seems like a nice way to debug. Regarding that, how would
 the impact be affected running it on asterisk box? I guess only port 5060
 is not too bad


 On 16 January 2014 14:09, Gareth Blades 
 mailinglist+aster...@dns99.co.ukwrote:

   On 16/01/14 10:47, Tiago Geada wrote:

  Hi folks,

  We've been having a weird issue... It is happening more often in the
 last few months...

  Most inbound calls, we have in our dialplan before Queue():

  Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL})
 ;

  So when the call rings a member, softphone will show this string 

  The issue is that sometimes the string showing in the softphone is not
 the same. Its a string from a past call, in the latest case I've seen, from
 about 40 days ago!!

  User took a screenshot, I've searched for that uniqueid showing in
 softphone in cdr, and that string was valid for a different call 40 days
 ago!!


  I searched full log, and Set() sets the correct string... I can't
 figure why softphone shows a string from a past call !!

  :(

  Any hints ?


   I would leave tcpdump running capturing port 5060 so you can load it
 onto wireshark and have a look at the sip headers. That will tell you if
 the SIP is incorrect or if its a problem with the client.

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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

Hi,

I transfered the capture to my local machine and opened it in wireshark, I
can search from there:
-- SIP Display info:
Sapo:0:243709253:1389884558.292163:SIP/covilha-pstn-000201f3

but I will add your comment to my notes.


I've already searched the asterisk FULL log, and seen the Set() line ..
shows the correct string, that should have been displayed on softphone ...




On 16 January 2014 15:25, Gareth Blades mailinglist+aster...@dns99.co.ukwrote:

  The SIP trace will give you an idea is perhaps something is becoming
 corrupted. If you keep a log of the asterisk console output (asterisk
 -rvvv) then you will see what it attempts to set the callerid to and any
 errors.

 Another tip. When you have a look at the sip trace you will see the
 call-id. If you make a note of this and run the following replacing the
 call-id and the trace file with the appropriate values it will display the
 sip trace in a very nice human readable format. tshark comes with the
 wireshark pakage and ngrep is part of epel repository if you are running
 centos.

 tshark -t ad -r '$tracefile' -R 'sip.Call-ID contains $callID' -w - |
 ngrep -I - -W byline -t



 On 16/01/14 14:57, Tiago Geada wrote:

  Second thought, that would only allow me to know if there is a problem
 on asterisk or softphone.

  Because the old callerid(name) that was presented on the softphone,
 belonged to a call made to a different peer, I doubt that it would be a
 softphone problem.

  Our softphones are fixed with the same peer/extension .. if the wrong
 callerid were originally called to the same peer.. I guess that would be
 worth it..

  even so, I will try and measure the impact on performance, however if
 asterisk did send the wrong string, how could I debug that??


 On 16 January 2014 14:27, Tiago Geada tiago.ge...@gmail.com wrote:

  You're right, seems like a nice way to debug. Regarding that, how would
 the impact be affected running it on asterisk box? I guess only port 5060
 is not too bad


  On 16 January 2014 14:09, Gareth Blades 
 mailinglist+aster...@dns99.co.uk wrote:

On 16/01/14 10:47, Tiago Geada wrote:

  Hi folks,

  We've been having a weird issue... It is happening more often in the
 last few months...

  Most inbound calls, we have in our dialplan before Queue():


 Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL})
 ;

  So when the call rings a member, softphone will show this string 

  The issue is that sometimes the string showing in the softphone is not
 the same. Its a string from a past call, in the latest case I've seen, from
 about 40 days ago!!

  User took a screenshot, I've searched for that uniqueid showing in
 softphone in cdr, and that string was valid for a different call 40 days
 ago!!


  I searched full log, and Set() sets the correct string... I can't
 figure why softphone shows a string from a past call !!

  :(

  Any hints ?


   I would leave tcpdump running capturing port 5060 so you can load it
 onto wireshark and have a look at the sip headers. That will tell you if
 the SIP is incorrect or if its a problem with the client.

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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

Looking at his tcpdump command it keeps 500 files of 10 MB each? (not sure)


On 16 January 2014 15:29, Kevin Larsen kevin.lar...@pioneerballoon.comwrote:

 asterisk-users-boun...@lists.digium.com wrote on 01/16/2014 08:55:31 AM:

  From: Gareth Blades mailinglist+aster...@dns99.co.uk
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com,
  Date: 01/16/2014 08:55 AM
  Subject: Re: [asterisk-users] Weird issue with
 Set(CALLERID(name)=string);
  Sent by: asterisk-users-boun...@lists.digium.com
 
  Very little as the amount of data being captured is quite small. We
  have it running on our production servers which routinely handle a
  couple of hundred concurrent calls.
 
  This is the script we use to start off the capture. It uses rolling
  capture files so we will always have the last X number of capture
  logs. It works very well and we have a custom system which enables
  us to search for calls and request traces for them for when we have
  to diagnose problems.
 
  #!/bin/bash
  cd /var/lib/asterisk/siptraces
  DATE=`date +%Y%m%d%H%M%S`
  TRACEFILE=/var/lib/asterisk/siptraces/$DATE-siptrace.pcap
  nohup /usr/sbin/tcpdump -p -i eth0 -s 0 port 5060 -w $TRACEFILE -C 10 -W
 500 
 
 

 Not to derail the conversation, Gareth, but do you leave this running full
 time on your asterisk boxes or just turn it on when you are trying to track
 problems?

 On average, how far back can you go for looking at problems?
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

Gareth,

I had to disable the tcpdump process, has we were having sound quality
issues.

:-(


On 16 January 2014 15:35, Gareth Blades mailinglist+aster...@dns99.co.ukwrote:

  On 16/01/14 15:29, Kevin Larsen wrote:

 Not to derail the conversation, Gareth, but do you leave this running full
 time on your asterisk boxes or just turn it on when you are trying to track
 problems?

 On average, how far back can you go for looking at problems?


 Its normally running full time so if someone reports a problem with a call
 we can look at the logs and find out exactly what happened. We keep
 asterisk verbose logs for 3 months, sip traces currently for about a month,
 and uk-isup traces for a couple of weeks.

 Most carriers will do something similar. BT for example keep all of their
 SS7 signalling for 48 hours.

 Regards
 Gareth

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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

And we just figured that sound quality issues were not due to tcpdump ..
anyway sorry to troll this feed, and thank you for your sugestion



On 16 January 2014 16:57, Tiago Geada tiago.ge...@gmail.com wrote:

 Gareth,

 I had to disable the tcpdump process, has we were having sound quality
 issues.

 :-(


 On 16 January 2014 15:35, Gareth Blades 
 mailinglist+aster...@dns99.co.ukwrote:

  On 16/01/14 15:29, Kevin Larsen wrote:

 Not to derail the conversation, Gareth, but do you leave this running
 full time on your asterisk boxes or just turn it on when you are trying to
 track problems?

 On average, how far back can you go for looking at problems?


 Its normally running full time so if someone reports a problem with a
 call we can look at the logs and find out exactly what happened. We keep
 asterisk verbose logs for 3 months, sip traces currently for about a month,
 and uk-isup traces for a couple of weeks.

 Most carriers will do something similar. BT for example keep all of their
 SS7 signalling for 48 hours.

 Regards
 Gareth

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Re: [asterisk-users] Caller ID not real nor showing in call logs.

[Tiago Geada]

logs ?

full log containing the call?


On 8 January 2014 14:56, Eherr email.eherr9...@gmail.com wrote:

 I have a multi tenant asterisk box where on tenant is receiving calls from
 the caller ID as1as and they cannot pickup the call.

 The caller ID also does not show up in the call log.

 Thoughts?

 Thanks,
 --Eric
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Re: [asterisk-users] Setting CDR variables for all linked channels

[Tiago Geada]

not sure about dial, but I Set(__var=value); and in each piece of dialplan
I set CDR(var=value);


On 31 December 2013 00:00, Igor Katson igor.kat...@gmail.com wrote:

 Hi,

 when one does Set(CDR(var)=value) in dialplan, the value is only set for
 one record in the cdr table, but not the linked ones (the ones with the
 same linkedid).
 E.g. if you do something like
 same = n, Set(CDR(var)=value)
 same = n,Dial(Local/somethingLocal/something2)

 like only the original CDR record with have var set to value, but the
 ones created from Dial won't.

 Is it possible to set the CDR variables in all the linked channels?

 P.S. And is it possible to find out by the CDR logs, if the originating
 call is in progress?

 Thanks!

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Re: [asterisk-users] Ctrl-W killing entire line, not just last word

[Tiago Geada]

I would guess you need to recompile ?


On 12 December 2013 20:07, Dotan Cohen dotanco...@gmail.com wrote:

 On Wed, Dec 11, 2013 at 10:20 PM, Tzafrir Cohen
 tzafrir.co...@xorcom.com wrote:
  You need libedit-dev, not libeditline-dev.
 

 Thank you Tzafrir. However, even after installing libedit and
 libedit-dev, Ctrl-W still kills (deletes) to the beginning of the
 line.


 --
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 http://gibberish.co.il
 http://what-is-what.com

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Re: [asterisk-users] Disable peer from AMI

[Tiago Geada]

I'm just stating what is already explained above. You could either have
dialplan with iftime() or use realtime peers, and have something
enable/disable them from sql backend


On 23 October 2013 11:38, Darryl Moore dar...@moores.ca wrote:

 put it in a different context in your dial plan and use a gotoif statement
 to control the times it is allowed to dial out. you can also redirect it to
 a prerecorded message whenever someone tries to use it during the 'off'
 time. no need for anything as brutal as disabling it in sip.conf.
 On 2013-10-23 12:37 AM, Michelle Dupuis mdup...@ocg.ca wrote:

  I need to disable/enable a peer after hours automatically, and am
 thinking about doing so via the AMI.

 Is there a command to enable/disable (or perhaps delete/add) a peer via
 the AMI?  I could create code to modify sip.conf and force a reload, but
 that seems like the wrong approach...

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Re: [asterisk-users] What linux distro most popular for Asterisk

[Tiago Geada]

debian wheezy compiled asterisk from source


On 18 October 2013 00:27, Andrew Furey andrew.fu...@gmail.com wrote:

 [Apologies, top-posting, Gmail, yadda yadda]

 As with a lot of software, I suspect the best answer is whichever distro
 YOU are most comfortable with. You're the one who has to support it, after
 all... Just my 2c.

 Andrew


 On Thursday, 17 October 2013, Rusty Newton wrote:

 On Tue, Oct 15, 2013 at 11:58 PM, Michelle Dupuis mdup...@ocg.ca wrote:
  Is there a recent survey of that Linux distro and version people are
 using
  for the Asterisk installations?  I recall seeing a pie chart over a
 year ago
  (I think on a wiki but I can't find it again)also hoping for
 something
  more current.
 
  I suspect RH5 and RH6 are most popular...but I'm looking for facts

 I don't have any numbers, but I watch the issue tracker a lot and I
 see pretty much CentOS, Debian and Ubuntu. Which seems to match what
 everyone else is saying on this thread.

 --
 Rusty Newton
 Digium, Inc. | Community Support Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 direct: +1 256 428 6200

 Check us out at: http://digium.com  http://asterisk.org

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 Language, be it English or something else, is the only tool flexible
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   -- Bill Garrett

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Re: [asterisk-users] Capture Media IP in CDR (CDR)

[Tiago Geada]

Hi,

I also doubt that the IP would do any good, anyway you store whatever you
want in your cdr, just Set(CDR(something)=${SIP_HEADER(Contact)}); and then
have the field something in your cdr storage


On 13 October 2013 21:25, jg webaccou...@jgoettgens.de wrote:

 I doubt that a media IP would really help, because there are proxies out
 there. If you need this kind of monitoring, then there are probably better
 ways to take care of this and they are independent of Asterisk.

 What you could do is to tap any traffic in the background, e.g. with
 tcpdump using the -G option and automatically delete the files after a
 certain period, unless there is a reason to keep the data. The pcap trace
 would contain a lot of relevant information, even if the traffic is
 encrypted (like timing data). Depending on national or local laws this
 might be even a more serious crime than threatening a school. It could
 still be justified to tap the traffic, like it is for other public
 authorities, but you would have to find out yourself whether you are or the
 school is allowed to do this.

 Actually, I tend to think that it is the school's task to enforce a
 specific security and surveillance concept and this also applies
 particularly to their IT structure. You are certainly not in the position
 to decide whether you should monitor anything unless it is part of your
 contract.

 Besides this, it is easy to store any kind of information along with
 classical CDR data. Just search for adaptive ODBC, or read the Asterisk
 book.

 jg


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Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

[Tiago Geada]

Hi,

Seems a great workaround from Gareth Blades. Thanks I will try it.

Any way to make asterisk log a line in /var/log/messages ?


On 10 October 2013 19:44, Michelle Dupuis mdup...@ocg.ca wrote:

  Gareth:

 Did you check if your message (or security) log recorded anything during
 these attempts?  If so, can you post the content of the logs during this
 attack?

 M
  --
 *From:* asterisk-users-boun...@lists.digium.com [
 asterisk-users-boun...@lists.digium.com] On Behalf Of Asghar Mohammad [
 asghar...@gmail.com]
 *Sent:* Tuesday, October 01, 2013 11:53 AM
 *To:* Asterisk Users List
 *Subject:* Re: [asterisk-users] Failed to authenticate user
 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

   Hi,
 Bad boys trying to guess a valid username.
 in sip.conf uncomment  alwaysauthreject=yes and Asterisk always reject 1st
 invite.


 On Tue, Oct 1, 2013 at 5:26 PM, Gareth Blades 
 mailinglist+aster...@dns99.co.uk wrote:

 On 01/10/13 15:44, gincantalupo wrote:

 On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo 
 gincantal...@fgasoftware.com wrote:

 Hi,

 I get a lot of these messages on my Asterisk CLI:

 Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS
 ;tag=03f82bb9

 as if my PBX machine is trying to authenticate to itself. It seems
 someone is attacking my asterisk PBX.

 Is there a way to fix this problem?


 in sip.conf I have guest connections permitted and have them going to the
 default context which contains :-

 [default]
 ; all unauthenticated connection attempts from the internet come in here.
 exten = _[+*#0-9].,1,NoOp(Unauthenticated call attempt -
 ${SIP_HEADER(Contact)})
 exten = _[+*#0-9].,n,Congestion

 Then in fail2ban I have it match the following :-

 failregex = Registration from .* failed for \'HOST\' - Wrong password
 Unauthenticated call attempt .*\@HOST\:


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Re: [asterisk-users] SIM adaptor (huwewi or other)

[Tiago Geada]

Hi,

We've used https://code.google.com/p/asterisk-chan-dongle/ in the past with
success, only one call per sim


On 29 September 2013 09:39, bilal ghayyad bilmar...@yahoo.com wrote:




   On Wednesday, September 11, 2013 1:54 PM, longst longst...@gmail.com
 wrote:
  I think GoIP gsm gateway also is a good choice

 Sent from Shitian Long


 On Sep 11, 2013, at 12:29 PM, bilal ghayyad bilmar...@yahoo.com wrote:

 Hello;

 I am looking for SIM adaptor to be connected with Asterisk to be able to
 send and receive calls from the mobile operator and if possible the same
 adapter to be used for SMS sending and receiving.

 But what if anyone called this SIM card that is connected to this adapter
 and no one relied his call, how this miss call can reach for the use at the
 asterisk PBX?

 Regards
 Bilal

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Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Tiago Geada]

Hi,

Our queue members are Local channels, thus when dialing the agent, the
dialplan will do several stuff including:

Set(CALLERID(name)=${CALLERID(name)}:Sales)
UserEvent(something,data: ${bunch-of-data-in-some-format})
Dial(SIP/final-agent-phone,timeout,A(Sales))

The UserEvent will be picked up by our client-register-ticket-stuff software

The announcement A() will be heard by the agent upon answering the call
like sales call


On 4 August 2013 02:59, Mitch Claborn mitch...@claborn.net wrote:

 We do something very similar.

 Use the gosub parameter of the Queue application to call a subroutine in
 the dial plan when the agent answers the call.

 same =n,Queue(sales,tc,,sub-**QueueConnected)

 [sub-QueueConnected]
 ; this runs on the agent/member's channel
 exten =s,1,NoOp()
   ; whatever you need to do here
   same =n,Return()

 See https://wiki.asterisk.org/**wiki/display/AST/Asterisk+11+**
 Application_Queuehttps://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue


 Mitch


 On 08/03/2013 12:45 PM, Timothy Smith wrote:

 Hello Folks,

 I am setting up a call center but we have few agents so one agent is
 able to handle calls of different languages and different queues. For
 the agent to identify the caller, I want a popup to appear as the
 phone starts to ring with the caller's number, language (selected in
 the IVR), Queue (sales, support etc) and any other information (e.g a
 URL with parameters)

 I can send this information either via netcat (to a client such as
 yac) to a Windows PC but the problem is I do not know when the caller
 is about to be connected to the agent, so that I run the command. If I
 wasn't using queues, it would be easy because  I would run the netcat
 command and then dial the user's extension.

 My Question is: Is there a way I can know when the caller is just
 about to be connected to an agent (when the agent's SIP extension
 starts ringing)?

 There are these settings setinterfacevar, setqueueentryvar,
 setqueuevar in queues.conf but when can I use them?

 Have you guys been in this situation before? Any alternative solutions
 (sending caller info to an agent)?

 I am using Asterisk 11 and Windows 7 PCs for agents.

 Thank you!

 Kind Regards,
 Wilson

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Re: [asterisk-users] Queues: Knowing when a caller is position 1 (agent phone ringing)

[Tiago Geada]

Hi,

You just said you use Local channels. Local channel is a dialplan that has
a Dial() to a sip device?

We use queues, and have a queue-macro that sends the UserEvent upon
bridging the call...


On 4 August 2013 16:41, Timothy Smith timotsm...@gmail.com wrote:

 Dear Tiago,

 Thanks for your answer, but I have a few questions.

 Do you use queues? We are operating a call centre with several queues,
 so I don't see how we would use the Dial command. When a call comes
 in, we enter the caller (depending on what options he has selected)
 into a queue. Do you have any alternative method, which would involve
 dialling the agent directly as you described below?

 regards,
 T

 On Sun, Aug 4, 2013 at 3:47 PM, Tiago Geada tiago.ge...@gmail.com wrote:
  Hi,
 
  Our queue members are Local channels, thus when dialing the agent, the
  dialplan will do several stuff including:
 
  Set(CALLERID(name)=${CALLERID(name)}:Sales)
  UserEvent(something,data: ${bunch-of-data-in-some-format})
  Dial(SIP/final-agent-phone,timeout,A(Sales))
 
  The UserEvent will be picked up by our client-register-ticket-stuff
 software
 
  The announcement A() will be heard by the agent upon answering the call
 like
  sales call
 
 
  On 4 August 2013 02:59, Mitch Claborn mitch...@claborn.net wrote:
 
  We do something very similar.
 
  Use the gosub parameter of the Queue application to call a subroutine in
  the dial plan when the agent answers the call.
 
  same =n,Queue(sales,tc,,sub-QueueConnected)
 
  [sub-QueueConnected]
  ; this runs on the agent/member's channel
  exten =s,1,NoOp()
; whatever you need to do here
same =n,Return()
 
  See
 
 https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue
 
 
  Mitch
 
 
  On 08/03/2013 12:45 PM, Timothy Smith wrote:
 
  Hello Folks,
 
  I am setting up a call center but we have few agents so one agent is
  able to handle calls of different languages and different queues. For
  the agent to identify the caller, I want a popup to appear as the
  phone starts to ring with the caller's number, language (selected in
  the IVR), Queue (sales, support etc) and any other information (e.g a
  URL with parameters)
 
  I can send this information either via netcat (to a client such as
  yac) to a Windows PC but the problem is I do not know when the caller
  is about to be connected to the agent, so that I run the command. If I
  wasn't using queues, it would be easy because  I would run the netcat
  command and then dial the user's extension.
 
  My Question is: Is there a way I can know when the caller is just
  about to be connected to an agent (when the agent's SIP extension
  starts ringing)?
 
  There are these settings setinterfacevar, setqueueentryvar,
  setqueuevar in queues.conf but when can I use them?
 
  Have you guys been in this situation before? Any alternative solutions
  (sending caller info to an agent)?
 
  I am using Asterisk 11 and Windows 7 PCs for agents.
 
  Thank you!
 
  Kind Regards,
  Wilson
 
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Re: [asterisk-users] Asterisk CPU use

[Tiago Geada]

I recently had high load average due to disk usage (IO) . I use
mixmonitor() to record to tmpfs and moved mysql to a different disk
(realtime, cdr etc).
Load average is now better.


On 31 July 2013 19:45, Paul Belanger paul.belan...@polybeacon.com wrote:

 On 13-07-29 10:22 AM, Eduardo Leones wrote:

 Hello, working in a call center where we set up a structure in asterisk.
 When my voip reaches 150 calls are with bad quality. We do not transcode
 codec. What I realized using the top command server (CentOS) processing is
 too high for the asterisk. But the general processor server is down. Would
 any limitation of Asterisk to use more hardware resources?


 Your load average is insane.  Time to off load resources from your PBX,
 for example why are you running httpd?  You need to figure out where your
 bottleneck is and then adjust it.

 Using something like iotop, netstat and see what your system is doing.

 I doubt this is a CPU issue.

 --
 Paul Belanger | PolyBeacon, Inc.
 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 Github: https://github.com/pabelanger | Twitter:
 https://twitter.com/pabelanger


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Re: [asterisk-users] Digitial Phones

[Tiago Geada]

Sorry, but what is a 'digital phone device' ?


On 14 July 2013 12:45, bilal ghayyad bilmar...@yahoo.com wrote:

 Hello;

 Does asterisk support Digital Phone devices? If yes, what is the required
 cards and in which channel to do the configuration? Is it dahdi or
 something else?

 In other words, the customer does not need IP Phones.

 Regards
 Bilal

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Re: [asterisk-users] Integration with Social Media, Email and Web call center

[Tiago Geada]

Hi,

We before, used facebook graph api (json) on a php script.
php would check new posts every minute, and write a new .call file into
asterisk, with a sort of TTS

call would go on queue, and once a member picks it up, he hears 'new
facebook call from, bla bla, stating bla bla bla'
He would then proceed to reply the facebook post (in our case also done in
our software that would post back to FB via graph api)


On 24 January 2013 15:28, Danny Nicholas da...@debsinc.com wrote:

 This is how I would see the process working
 1.  use curl/wget to query Facebook (etc.)
 2.  determine whether we are to drop a call into the queue or just process
 a
 message
 3.  determine agent availability through AMI process or asterisk -rx
 process.
 4.  drop the call into the queue or place the message if the agent is
 available
 5.  if the agent is unavailable, do alternate process.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal
 ghayyad
 Sent: Thursday, January 24, 2013 9:24 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Integration with Social Media, Email and Web
 call center

 They advised me to check jabber.org.
 Yes, jabber.org has a client that can send/receive and integrate with
 other
 social media (facebook, msn, twitter, ... etc).

 But, as an Agent who can login/logout and take a calls, how can I make it
 to
 be single login for voice and messages. So, if the agent is not available,
 he will not get a calls and will not get a messages.

 Those who used jabber.org or who used other than jabber.org for such
 requirement, what do you suggest?

 Regards
 Bilal

 --

 
  For just the messaging part, you should be able to use wget or curl to
  interface and create messages.  You might have to go a little higher
  level
  like C or Perl, but it sounds very doable.
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-boun...@lists.digium.com]
  On Behalf Of bilal ghayyad
  Sent: Tuesday, January 22, 2013 4:27 PM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] Integration with Social Media, Email and Web
  call center
 
  Dears;
 
  Can someone advise me where to find a technology (open
  source) that let us
  able to integrate with social media like whatsapp and facebook? And
  use this in call center (queuing the messages and routing it for
  agent)?
 
  Anyone give me a light to start?
 
  Regards
  Bilal

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Re: [asterisk-users] question on SIP trunk and AMI to place call

[Tiago Geada]

Have you tried and looked up all events generated when you place the call?

some of them are bound to have the variable callerid set


On 24 January 2013 16:46, Jerry Geis ge...@pagestation.com wrote:

 When I am monitoring the AMI I see the following event
 for a call I just made over a SIP trunk.

 Event: Newchannel
 Privilege: call,all
 Channel: SIP/testmachine-000d
 ChannelState: 0
 ChannelStateDesc: Down
 CallerIDNum:
 CallerIDName:
 AccountCode:
 Exten:
 Context: testmachine
 Uniqueid: 1359035395.20

 In this event or any event following I do not see
 the phone number that I dialled. How do I correlate
 the SIP/testmachine-000d to the number I just dialed
 (purpose is to hangup the call later if I need to interrupt it)

 Now if I am using a machine with actual hardware cards, the phone
 number is included as part of the Channel so I can look that up.
 but for a SIP trunk the phone number dialled does not come over the AMI.

 How do I match up the call I just started (using AMI over SIP trunk) to
 the number I called?

 Thanks,

 jerry



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Re: [asterisk-users] SIP and RTP on different IP's

[Tiago Geada]

yes I have no control over that.

Ok we will figure another way. Thanks


On 25 November 2012 07:10, Duncan Turnbull dun...@e-simple.co.nz wrote:



 On 25/11/2012, at 1:23 PM, Tiago Geada tiago.ge...@gmail.com wrote:

 linux does sort this out and asterisk listens in both interfaces. however
 asterisk connects and tells remote end to send rtp back at the same IP
  where sip is going trough...

 remote end does try to send  it but gets stopped in a firewall.. thus if
 asterisk did present a different  IP to recieve RTP in its SIP header, this
 would not happen!



 I think this is outside of asterisk's natural ability

 You may need a proxy server in between you and the Cisco to achieve this
 if you can't change the firewall.

 http://forums.asterisk.org/viewtopic.php?f=1t=84018

 Have you tried making the preferred route to these addresses go out eth1,
 thus getting the required address?

 Ultimately seems odd the firewall allows access in but not out, guessing
 you have no control over that?

 Good luck

 Cheers Duncan


 On 23 November 2012 19:39, Duncan Turnbull dun...@e-simple.co.nz wrote:


 On 24/11/2012, at 2:19 AM, Tiago Geada tiago.ge...@gmail.com wrote:

 Hello Folks, I am looking for a way that makes asterisk tell remote SIP
 party that the IP where they will send RTP is not the same as the one I am
 comunicating via SIP

 Can this be done anyhow?

 I can try and explain:

 We have placed a asterisk box in our partners office.

 It has eth0 with IP 172.16.1.10 and eth1: 10.34.18.250

 linux has its routes set so it can comunicate with several networks in
 their offices.

 now there is a cisco call manager that we need to communicate with.
 Normally via our IP 172.16.1.10, however seems that this cisco uses some
 sort of 'directmedya=yes' and sets both ends speaking RTP with themselves.

 There are some extensions in cisco that have a network 10.134.0.0/16that we 
 can only comunicate via eth1

 thus when calling cisco (always via eth0) sometimes we need to say that
 OUR IP to recieve RTP is not 172.16.1.10, but 10.34.18.250


 This is a routing issue, not asterisk I think. You are saying you route
 to cisco via eth0, it sets up connections to its end points and then drops
 out of the media flow, but the end points have no route to the eth0 address
 so they fail

 Linux usually sorts this out and asterisk replies on the address of the
 interface it sends out with. So for the most part the response in my
 experience if its going out eth1 should use the eth1 ip address.

 If you can get to it via eth0 and thats the preferred route then it will
 have the eth0 address. If so why can't you change your routing table to use
 eth1 when you need to go to the cisco then you will have the right address
 and the far extensions can respond to you correctly

 Or change the cisco network endpoints so they can successfully access
 your address on eth0


 can this be done?
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Re: [asterisk-users] How to tie orders taken to specific CDR records

[Tiago Geada]

Hello,

I faced this issue a while back. What I do, as soon the call comes in, I
Set(UID=${UNIQUEID}), then re-use UID allong the dialplan as
Set(CDR(UID)=${UID}).

My DB has a UID field, that I can group by

On 27 October 2012 10:26, Bharat Lalcheta bharatlalch...@gmail.com wrote:

 Its depends on dialplan and the way you treat the call.


 On Fri, Oct 26, 2012 at 7:54 PM, Mitch Claborn mitch...@claborn.netwrote:

 Looking at the uniqueid, I get multiple records for some of them.  Am I
 getting more than one CDR record per call in some cases?

 SELECT uniqueid, COUNT(*) FROM asterisk_cdr
 GROUP BY uniqueid
 HAVING COUNT(*)  2


 Mitch


 On 10/26/2012 08:34 AM, Bharat Lalcheta wrote:


 Every CDR has uniqueid/callid generated and unique between all records.
 This callid generated when call arrives on system. And logged in CDR
 record as well. You can use it in your dialplan to bind with your order
 like
 exten = s,1,Set(ORDERID=${UNIQUEID})


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[asterisk-users] asterisk 1.8 realtime queue_log

[Tiago Geada]

Hello!!

is there a way to make asterisk 1.8 record queue_log in MySQL in the same
structure as asterisk 1.6 did?

column time was always inserted in UNIX TIME STAMP format

column data had all the data separated with pipes |

Is it possible to keep the same structure on 1.8 ???
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Re: [asterisk-users] Requiring agent to confirm queue calls only when forwarded to external device

[Tiago Geada]

forward to a Local extension that has dialplan requiring the
acknowledgement?

On 16 August 2012 21:12, Phil Frost p...@macprofessionals.com wrote:

 I'd like to allow my users to forward their calls using the forwarding
 feature on their SIP handsets and continue to receive Queue() calls.
 Currently I set the 'i' option in Queue() so that if a user forwards to
 their cell phone, or any other extension that has voicemail, the voicemail
 doesn't eat all the calls to the queue.

 I'm aware that I can configure the queue to require agents to acknowledge
 the call. However, most of the calls go to internal devices where
 confirmation isn't necessary, so I'd like to avoid the extra inconvenience
 in that most common case.

 What I'd like to do is somehow detect that a handset has responded with a
 SIP 302 response, and only when this is the case, require the agent to
 confirm humanness before answering the call from the queue. Any ideas on
 how this could be implemented?


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Re: [asterisk-users] Unclosed channel

[Tiago Geada]

try the dial option 'g' that carries on with dialplan

On 8 June 2012 09:26, Khaled W. Chehab kche...@xplorium.com wrote:

 Dears,



 My scenario is to accept the call from user àAnswer the call -àplay mohà
 dial(SIP/Trunk,X)

 The problem is when the user send the bye the trunk call will not hangup

 How to solve this issue





 exten = 446696,1,Ringing

 exten = 446696,n,Answer()

 exten = 446696,n,Wait(2)

 exten = 446696,n,Playback(Welcome)

 exten = 446696,n,Dial(SIP/Trunk/${EXTEN},300)

 exten = 446696,n,Hangup





 How to solve such issue

 Thanks in advance


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Re: [asterisk-users] Deleting OLD Voicemails

[Tiago Geada]

I use find on a cron schedule to remove old recordings everyday. Im sure
you can do the same

find -H /var/log/asterisk/asterisk_rec/* -mtime +90 -type f -exec rm -v {}
\;

anything older than 90 days

On 27 May 2012 09:20, Eric Wieling ewiel...@nyigc.com wrote:


 I believe one of the patches involved in fixing for The Great Voicemail
 Problem* about a year ago was to make voicemail automatically renumber the
 mailbox files if it saw a gap.

 * from memory: The Great Voicemail Problem is a bug where if you received
 a new voicemail while listening to a message, the mailbox was not
 renumbered correctly when you deleted a message.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias
 Sent: Saturday, May 26, 2012 10:54 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Deleting OLD Voicemails

 I did not understand. What do you mean with renumber all the messages?

 El 25/05/2012 02:27, Edwin Lam edwin@officegeneral.com escribió:


On 5/23/12 2:42 AM, Danny Dias wrote:


Can i delete like this:

rm -rf
 /var/spool/asterisk/voicemail/voicemailcontextcustomer/300/INBOX/*.*

Is that ok? will this break something?



that's ok
no it shouldn't break anything.
however if you're going to delete some of the messages. you have to
renumber all the messages so that they are consecutive otherwise
the voicemail application may give you grief.



A little doubt here, once the user hears the voicemail
 using the phone, the
message is automatically moved to Old folder, is that right?



yes


--
Edwin Lam edwin@officegeneral.com
Systems Engineer, OfficeWyze, Inc.
 Ph: +1 415 439 4988 tel:%2B1%20415%20439%204988  Fax: +1 415
 283 3370 tel:%2B1%20415%20283%203370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 
 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20


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Re: [asterisk-users] hangup not detected?

[Tiago Geada]

Looks like Swift() (whatever that is) is not returning ?

On 24 May 2012 23:07, Justin Killen jkil...@allamericanasphalt.com wrote:

 ** ** **

 Here is the output from the cli:

 ** **

 dozer*CLI core show channels

 Channel  Location State   Application(Data)

 DAHDI/5-1s@DB_LOOKUP:24   Up  Swift(Schedule for
 employee

 1 active channel

 1 active call

 1528 calls processed

 dozer*CLI core show channel dahdi/5-1

  -- General --

Name: DAHDI/5-1

Type: DAHDI

UniqueID: 1337821128.1363

LinkedID: 1337821128.1363

   Caller ID: (N/A)

  Caller ID Name: (N/A)

 Connected Line ID: (N/A)

 Connected Line ID Name: (N/A)

 DNID Digits: (N/A)

Language: en

   State: Up (6)

   Rings: 1

   NativeFormats: 0x4 (ulaw)

 WriteFormat: 0x4 (ulaw)

  ReadFormat: 0x4 (ulaw)

  WriteTranscode: No

   ReadTranscode: No

 1st File Descriptor: 15

   Frames in: 3967

  Frames out: 15882

  Time to Hangup: 0

Elapsed Time: 20h56m23s

   Direct Bridge: none

 Indirect** **Bridge: none

  --   PBX   --

 Context: DB_LOOKUP

   Extension: s

Priority: 24

  Call Group: 0

Pickup Group: 0

 Application: Swift

Data: Schedule for employee number :  Thursday, May
 24th, 2012, you are scheduled at XX

 Blocking in: (Not Blocking)

   Variables:

 READSTATUS=TIMEOUT

 return_id=

 MAX_REPEAT=4

 ODBCSTATUS=SUCCESS

 ODBCROWS=1

 COUNTER=2

 AAA_OUTPUT=Schedule for employee number :  Thursday, May 24th, 2012,
 you are scheduled at XX..

 data=Thursday, May 24th, 2012, you are scheduled at XX

 id=

 ODBC_FETCH_STATUS=SUCCESS

 ~ODBCFIELDS~=id,data

 ODBC_ID=903

 ID_VALIDATED=AAA_VALIDATE_EMP_NUM(27,)

 account_id=

 read_length=7

 get_param2=E

 get_param1=27

 validate_func=AAA_VALIDATE_EMP_NUM

 truck_text=employee number

 readprompt=AAA/enter_employee_number

 comp_num=27

 BACKGROUNDSTATUS=SUCCESS

 ** **

   CDR Variables:

 level 1: dnid=

 level 1: dst=4

 level 1: dcontext=default

 level 1: channel=DAHDI/5-1

 level 1: lastapp=Swift

 level 1: lastdata=Schedule for employee number :  Thursday, May
 24th, 2012, you are schedu

 level 1: start=2012-05-23 17:58:48

 level 1: answer=2012-05-23 17:58:54

 level 1: duration=75383

 level 1: billsec=75377

 level 1: disposition=ANSWERED

 level 1: amaflags=DOCUMENTATION

 level 1: accountcode=27_EMP

 level 1: uniqueid=1337821128.1363

 level 1: linkedid=1337821128.1363

 level 1: userfield=2885

 level 1: sequence=1363

 ** **

 ** **

 ** **

 ** **

 ** **

 Since the ‘lastapp’ variable is ‘Swift’, this would indicate that the
 cepstral wrapper is having a problem, correct?

 ** **

 Justin Killen 
   --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Justin Killen
 *Sent:* Tuesday, May 22, 2012 8:53 AM

 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] hangup not detected?
 

  ** **

 Okay, the next time it gets in this state I’ll gather that information.***
 *

 ** **

 Justin Killen
   --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren Selby
 *Sent:* Monday, May 21, 2012 1:22 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] hangup not detected?

 ** **

 On Fri, May 18, 2012 at 12:00 PM, Justin Killen 
 jkil...@allamericanasphalt.com wrote:

 I have and automated call-in dispatch system where hundreds of people call
 in daily for 2-3 minutes each.  The extension is set up to get their
 information, then text-to-speech the dispatch information (via odbc).  It
 then loops 5 times then ends the call.  These calls are being handled by an
 8 port analog digium card.  

  

 Sometimes though, I see calls via ‘core show channel dahdi/1-1’ that have
 a time of  16 hours.  I’m not sure if this is a result of dahdi missing
 the hangup, ODBC timing out, or TTS failing for some reason.  When a
 channel gets in this state, the call doesn’t seem to progress through the
 dialplan, they always display the TTS line.  Doing a ‘dahdi destroy channel
 1-1’ doesn’t seem to be effective – the only way I’ve been able to clear
 the calls is to do a ‘dahdi restart’ and/or restart the asterisk service.*
 ***

  

 For TTS I’m using cepstral with the Swift wrapper.

  

 Here is a snippet of my 

Re: [asterisk-users] MixMonitor and ChanSpy

[Tiago Geada]

that means that from 1.4.18 that issue is no longer present ?

On 7 February 2012 12:44, Jonas Kellens jonas.kell...@telenet.be wrote:

 **
 On 02/07/2012 01:07 PM, Sammy Govind wrote:

 Hello,

  I've been managing multiple call centres, almost all of them having
 their calls recorded one way or other. Even in PBX environments with
 MixMonitor and call recordings I haven't came across the situation where I
 discovered that I can't chanspy a call because its recorded !
 Which version of asterisk you are using ! can you paste the CLI logs which
 show a complete call with a failed attempt to Chanspy ?


 Using Asterisk 1.6.2.22.

 The fact that ChanSpy can not be used with MixMonitor is something I read
 on the wiki :

 Attention

- Up to and including Asterisk 1.4.17 ChanSpy can cause a *
crash/segfault* if used together with 
 Monitorhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Monitoror
MixMonitor http://www.voip-info.org/wiki/view/MixMonitor at the same
time. 1.4.18 is supposed to attack this issue by using audiohooks that
replaces the current ChanSpy approach.



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Re: [asterisk-users] Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted

[Tiago Geada]

use 'ulimit' to set a higher value on max open file descriptors

On 2 July 2011 02:00, Eric Wieling ewiel...@nyigc.com wrote:



  -Original Message-
  From: asterisk-users-boun...@lists.digium.com
  [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
  Kaushal Shriyan
  Sent: Friday, July 01, 2011 8:28 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Starting asterisk:
  /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot
  modify limit: Operation not permitted
 
  On Fri, Jul 1, 2011 at 11:13 AM, Kaushal Shriyan
  kaushalshri...@gmail.com wrote:
   Hi
  
   Please help me understand about the below issue ?
  
   [root@asterisk1 ~]# /etc/init.d/asterisk restart Stopping
   safe_asterisk:[  OK  ] Shutting
   down asterisk:[  OK  ] Starting
   asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open
   files: cannot modify limit: Operation not permitted
 [  OK  ]
   (reverse-i-search)`d': /etc/init.d/asterisk restart
   [root@asterisk1 ~]# rpm -qa | grep asterisk
   asterisk-sounds-core-en-gsm-1.4.21-1_centos5
   asterisk18-1.8.4.4-1_centos5
   asterisk18-core-1.8.4.4-1_centos5
   asterisk18-doc-1.8.4.4-1_centos5
   asterisk18-dahdi-1.8.4.4-1_centos5
   asterisk18-configs-1.8.4.4-1_centos5
   asterisk18-voicemail-1.8.4.4-1_centos5
   [root@asterisk1 ~]# uname -a
   Linux asterisk1 2.6.18-238.el5 #1 SMP Thu Jan 13 15:51:15 EST 2011
   x86_64 x86_64 x86_64 GNU/Linux
   [root@asterisk1 ~]# cat /proc/version
   Linux version 2.6.18-238.el5 (mockbu...@builder10.centos.org) (gcc
   version 4.1.2 20080704 (Red Hat 4.1.2-48)) #1 SMP Thu Jan
  13 15:51:15
   EST 2011
   [root@asterisk1 ~]# cat /etc/redhat-release CentOS release
  5.6 (Final)
   [root@asterisk1 ~]#
  
   Regards
  
   Kaushal
  
 
  Hi Again,
 
  Can someone please reply on my earlier post to this emailing list.

 This is an operating system question.  The link is for core size, but the
 basic concept should work for open files as well.


 http://superuser.com/questions/79717/bash-ulimit-core-file-size-cannot-modify-limit-operation-not-permitted

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Re: [asterisk-users] call files .vbs

[Tiago Geada]

I would rather write a new bash script for text and file handing.

I think you can install MONO and run windows stuff... from .net to vbs

On 23 May 2011 08:09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:

 On Sun, May 22, 2011 at 07:05:45PM -0400, Thomas Perron wrote:
  This may be an obvious reflection of my Asterisk/Linux/Windows weaknesses
  but I want to know in any case!
 
  Can a vb script run somehow on a Linux machine or does it only work on
  Windows?

 Only on Windows (practically).

 
  If I were to build a call file script (described in this link
  http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out )
 then
  how does it work if my Asterisk machine is running on Centos 5.5?
 
  I simply want to execute a script that helps me automate the voice
  broadcasting/IVR of up to 1 phone numbers.

 I assume you know what you're doing and this is for a good cause.

 Use the Asterisk Manager Interface.
 http://www.voip-info.org/wiki/view/Asterisk+manager+API

 Specifically, the Originate command.

 --
   Tzafrir Cohen
 icq#16849755  jabber:tzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] how to know how many calls are on hold

[Tiago Geada]

core show channels concise

Those with '(None)' haven't been briged yet.

On 17 May 2011 15:16, virendra bhati virbh...@gmail.com wrote:

 hi list,

 please help me how to know how many calls are on hold.

 --



 -
 Thanks and regards

  Virendra Bhati
 +91-9172341457
 Asterisk Engineer


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[asterisk-users] estimated queue hold time

[Tiago Geada]

Hello list,

I'm looking for a way to have the estimated hold time on a queue prior to
joining it.

someone suggested to me to Queue() first for 1 sec, read
variable QUEUEHOLDTIME, validade it and Queue() again.

But as we're using real time configuration that would mean a event
ENTERQUEUE and a LEAVEQUEUE  too much in MySQL's queue_log


any suggestions??


Thanks in advance
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Re: [asterisk-users] how to know status of asterisk from php

[Tiago Geada]

The error is pretty straight forward. It is telling you that no Asterisk
service is running in that machine

On 28 April 2011 07:19, virendra bhati virbh...@gmail.com wrote:

 Hi,

 As per you suggestion I write small php scripts but didn't get result.
 Below is the php script and output of programs too.

 *PHP Script:-*

 ?php
 $priline = system('/usr/sbin/asterisk -rnx pri show spans',$pri);
 $asterisk = system(/etc/init.d/asterisk status, $asterisks);
 $mysql = system(/etc/init.d/mysql status,$mysqls);
 echo priline=.$priline;
 echo br;
 echo pri=.$pri;
 echo br;
 echo asterisk=.$asterisk;
 echo br;
 echo asterisks=.$asterisks;
 echo br;
 echo mysql=.$mysql;
 echo br;
 echo mysqls=.$mysqls;
 echo br;
 ?

 *Output:-*

 Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
 priline=Unable to connect to remote asterisk (does /var/run/asterisk.ctl
 exist?)
 pri=1
 asterisk=
 asterisks=127
 mysql=
 mysqls=127



 On Wed, Apr 27, 2011 at 8:43 PM, Juan David Diaz juanch...@gmail.comwrote:

 Hi:

 http://php.net/manual/en/function.system.php

 Then, the commands you shoul run:

 /usr/sbin/asterisk -rnxpri show spans
 /etc/init.d/asterisk status
 /etc/init.d/mysql status
 .
 .
 .
 .
 and so on!!

 good luck!

 Regards.

 Juan.
 Linux User #441131


 On Wed, Apr 27, 2011 at 6:22 AM, virendra bhati virbh...@gmail.comwrote:

 Hi

 How to know status of Asterisk,Mysql. PRI lines and other services from
 PHP scripts ?

 
 Thanks and regards

  Virendra Bhati
 +91-9172341457


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 -

 Thanks and regards

  Virendra Bhati
 +91-9172341457


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Re: [asterisk-users] Echocancellation OSLEC vs MG2 ?

[Tiago Geada]

linux-dahdi/README has a section on how to compile and install oslec

On 27 April 2011 22:15, Anthony Messina amess...@messinet.com wrote:

 On 04/27/2011 02:06 PM, satish patel wrote:
  Which echo cancellation is good between OSLEC and MG2. Dahdi by default
 use MG2 echo cancellation on channel.  If i want to use OSLEC then what
 should i need to do ? Do i need to recompile dahdi with OSLEC ?

 Yes, you would need to compile the OSLEC kernel module.  Or, if you are
 using a RedHat/Fedora based distro, you're welcome to use the
 dahdi-linux and dahdi-linux-kmod RPMS I build here.  I include OSLEC
 with the dahdi-linux-kmod build.

 http://messinet.com/rpms/

 --
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 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E


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Re: [asterisk-users] Best modem for chan_datacard

[Tiago Geada]

I used succesfully huawei E1550

On 24 April 2011 16:45, Dovid Bender asteriskus...@dovid.net wrote:

  Hi List,

 I am looking to play around with chan_datacard. Any advice on the best
 device to test with (that I can find on eBay) ?

 Regards,

 Dovid


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Re: [asterisk-users] call files

[Tiago Geada]

Hello,

Thanks for replying.

Answers below:

On 23 April 2011 18:29, Sherwood McGowan sherwood.mcgo...@gmail.com wrote:



 On Sat, Apr 23, 2011 at 11:20 AM, Tiago Geada tiago.ge...@gmail.comwrote:

 Hi.

 Im having trouble setting variables in channel dialplan and re-using them
 in Extension dialplan...

 Im using the following call file:

 Channel: Local/210332450@ZonNew-Outbound
 CallerID: ZonNew-Outbound:49:210332450:
 MaxRetries: 5
 RetryTime: 10
 WaitTime: 60
 Account: Outbound210332450
 Context: agents
 Extension: 888210332450
 Set: __PARTNER=ZonNew-Outbound
 Set: NUMBER=210332450


 -

 In  Local/210332450@ZonNew-Outbound I Set(bla='blabla');

 It seems I cannot re-use this var in extension _888X in context
 agents...


 Basically the Channel dialplan has a Queue() and in _888X I would
 like to know the peer (or interface) that answered it... What can I do?

 Thanks in advance


 I'm a little confused by It Seems I cannot re-use this var in extension
 _888XX in context agentsOf course you can use it...but if you
 set bla to a different value in your code where your callfile is processed,
 Asterisk will (rightfully so) just set bla = to whatever you set it to

 Now, if the callfile doesn't send a channel through the context that
 you're trying to set blah, that's a little odd...

 Now, as far as retrieving the information about the interface that answered
 the calllook in queues.conf.samplethere's a nifty configuration
 option:

 *setinterfacevar=no ; (the default is no)*

 Yes, I am aware of this and I do use it. However, I cannot use
MEMBERINTERFACE variable in dialplan _888X, and that is where I'm
needing it.

Also seems that its two channel legs and the only way would be to use
IMPORT() o SHARED() and for that I would have to know the channel name...

I am right now using IMPORT() like:

Set(CALLERID(num)=${IMPORT(${CHANNEL:0:$[${LEN(${CHANNEL})} -
1]}2,MEMBERNAME)});


but I fee that it is a ugly fix. What if call leg changes from 2 to 3?


 That option, when set to yes, causes several variables to be created *just
 * prior to the caller being bridged with the queue member...

 --
 Sherwood McGowan
 Telecommunications and VOIP Consultant


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[asterisk-users] call files

[Tiago Geada]

Hi.

Im having trouble setting variables in channel dialplan and re-using them in
Extension dialplan...

Im using the following call file:

Channel: Local/210332450@ZonNew-Outbound
CallerID: ZonNew-Outbound:49:210332450:
MaxRetries: 5
RetryTime: 10
WaitTime: 60
Account: Outbound210332450
Context: agents
Extension: 888210332450
Set: __PARTNER=ZonNew-Outbound
Set: NUMBER=210332450


-

In  Local/210332450@ZonNew-Outbound I Set(bla='blabla');

It seems I cannot re-use this var in extension _888X in context
agents...


Basically the Channel dialplan has a Queue() and in _888X I would
like to know the peer (or interface) that answered it... What can I do?

Thanks in advance
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Re: [asterisk-users] call files

[Tiago Geada]

Hi,

Using DumpChan(); Seems that Channel (where the call goes first) is a
sub-channel of Context/Extension (where the call goes on CONNECT) ??

first I have:
 Dumping Info For Channel: Local/210332450@ZonNew-Outbound-66c7;2:

Then after:
Dumping Info For Channel: Local/210332450@ZonNew-Outbound-66c7;1:

Help ?



On 23 April 2011 17:20, Tiago Geada tiago.ge...@gmail.com wrote:

 Hi.

 Im having trouble setting variables in channel dialplan and re-using them
 in Extension dialplan...

 Im using the following call file:

 Channel: Local/210332450@ZonNew-Outbound
 CallerID: ZonNew-Outbound:49:210332450:
 MaxRetries: 5
 RetryTime: 10
 WaitTime: 60
 Account: Outbound210332450
 Context: agents
 Extension: 888210332450
 Set: __PARTNER=ZonNew-Outbound
 Set: NUMBER=210332450


 -

 In  Local/210332450@ZonNew-Outbound I Set(bla='blabla');

 It seems I cannot re-use this var in extension _888X in context
 agents...


 Basically the Channel dialplan has a Queue() and in _888X I would
 like to know the peer (or interface) that answered it... What can I do?

 Thanks in advance

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Re: [asterisk-users] Voicemail to email issue

[Tiago Geada]

that is a sendmail issiue. Obviously asterisk is contacting 127.0.0.1 to try
and deliver e-mail.

Try help with sendmail folks, check that 127.0.0.1 is in the allowed to
relay list or so..

On 11 April 2011 21:11, satish patel satish...@hotmail.com wrote:

  Hi All,

 I have asterisk 1.8.3.2 and having issue with not getting VoiceMail email.
 I can send mail through command line using sendmail but not via asterisk. We
 have centralized zimbra email server. why its trying to send email to local
 127.0.0.1 address? is there any other configuration i am missing ?

 $cat voicemail.conf

 serveremail=aster...@shirley.example.com
 sendvoicemail=yes

 7623 = ,Satish Patel,sat...@example.com,,attach=yes|delete=yes



 $cat /var/log/mail.log

 Apr 11 12:57:57 shirley sendmail[29698]: p3BJvvtp029698: from=asterisk,
 size=9339, class=0, nrcpts=1,
 msgid=Asterisk-1-1388167162-7623-29658@shirley, relay=asterisk@localhost
 Apr 11 12:57:57 shirley sendmail[29698]: p3BJvvtp029698: to=Satish Patel
 sat...@example.com, ctladdr=asterisk (50011/50011), delay=00:00:00,
 xdelay=00:00:00, mailer=relay, pri=39339, relay=[127.0.0.1] [127.0.0.1],
 dsn=4.0.0, stat=Deferred: Connection refused by [127.0.0.1]


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[asterisk-users] One PRI card with 2 (or more) Telcos

[Tiago Geada]

Hi list!

We currently have a PRI gateway composed by a box with two Digium quad-span
PRI cards (a TE420 and a ).
One of the cards is filled with TELCO1, while the other has first two slots
filled with TELCO2, and 3rd slot with TELCO3.

I am currently having (timer ?) issues on TELCO3 (span 7)

D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing
on-going calls to terminate.
Problem clears immediately tho. I send a copy of the log with pri debug at a
time of problems...

Is there a problem having 2 telcos on the same PRI card?
Would somebody help?

asterisk*CLI pri show span 7
Primary D-channel: 202
Status: Provisioned, Up, Active
Switchtype: EuroISDN
Type: CPE
Overlap Dial: 0
Logical Channel Mapping: 0
Timer and counter settings:
  N200: 3
  N202: 3
  K: 7
  T200: 1000
  T202: 1
  T203: 1
  T303: 4000
  T305: 3
  T308: 4000
  T309: 6000
  T313: 4000
  T-HOLD: 4000
  T-RETRIEVE: 4000
  T-RESPONSE: 4000
Overlap Recv: No


and

[Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (I): T200
expired N200 times sending RR/RNR in state 8(Timer recovery)
[Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
[Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: Changing from state 8(Timer
recovery) to 5(Awaiting establishment)
[Mar 18 17:04:06] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
[Mar 18 17:04:07] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
[Mar 18 17:04:08] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
[Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (G): *T200
expired N200 times sending SABME in state 5(Awaiting establishment)*
[Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Changing from state
5(Awaiting establishment) to 4(TEI assigned)
[Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 DL event:
Q931_DL_EVENT_DL_RELEASE_IND(3)
[Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=56
on channel 2
[Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=64
on channel 3
[Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=58
on channel 4
[Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=66
on channel 6
[Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
[Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Changing from state 4(TEI
assigned) to 5(Awaiting establishment)
[Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c:   == Primary D-Channel on
span 7 down
[Mar 18 17:04:09] WARNING[19844] chan_dahdi.c: No D-channels available!
 Using Primary channel 202 as D-channel anyway!
[Mar 18 17:04:10] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
[Mar 18 17:04:11] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
[Mar 18 17:04:12] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
[Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (G): T200
expired N200 times sending SABME in state 5(Awaiting establishment)
[Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: Changing from state
5(Awaiting establishment) to 4(TEI assigned)
[Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 DL event:
Q931_DL_EVENT_DL_RELEASE_IND(3)
[Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
[Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: Changing from state 4(TEI
assigned) to 5(Awaiting establishment)
[Mar 18 17:04:13] WARNING[19844] chan_dahdi.c: No D-channels available!
 Using Primary channel 202 as D-channel anyway!
[Mar 18 17:04:14] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: T309 timed out waiting for
data link re-establishment
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: q931.c:7750
pri_internal_clear: Call 56 enters state 0 (Null).  Hold state: Idle
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: clearing, alive 1, hangupack
0
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c:   == Primary D-Channel on
span 7 up
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: -- Channel 0/2, span 7
got hangup, cause 27
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: T309 timed out waiting for
data link re-establishment
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: q931.c:7750
pri_internal_clear: Call 64 enters state 0 (Null).  Hold state: Idle
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: clearing, alive 1, hangupack
0
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: -- Channel 0/3, span 7
got hangup, cause 27
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: T309 timed out waiting for
data link re-establishment
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: q931.c:7750
pri_internal_clear: Call 58 enters state 0 (Null).  Hold state: Idle
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: clearing, alive 1, hangupack
0
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: -- Channel 0/4, span 7
got hangup, cause 27
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: T309 timed out waiting for
data link re-establishment
[Mar 18 17:04:15] VERBOSE[19844] chan_dahdi.c: q931.c:7750
pri_internal_clear: Call 66 enters state 0 (Null).  Hold 

Re: [asterisk-users] One PRI card with 2 (or more) Telcos

[Tiago Geada]

Just a follow up with a bit more information

asterisk*CLI module show like timing
Module Description  Use
Count
res_timing_pthread.so  pthread Timing Interface 0

*res_timing_dahdi.soDAHDI Timing Interface
  40*
2 modules loaded
asterisk*CLI


--

 [root@asterisk ~]# dahdi_test -c 100

Opened pseudo dahdi interface, measuring accuracy...

99.999% 99.999% 99.992% 99.997% 99.998% 99.995% 99.998% 99.996%

99.997% 99.998% 99.997% 99.994% 99.991% 99.999% 99.998% 99.998%

99.995% 99.993% 99.998% 99.999% 99.998% 99.995% 99.992% 99.998%

100.000% 99.998% 99.995% 99.992% 99.999% 99.998% 99.998% 99.999%

99.995% 99.999% 99.999% 99.998% 99.999% 99.997% 99.999% 99.998%

99.998% 99.996% 99.992% 99.998% 99.998% 99.999% 99.996% 99.992%

99.999% 99.998% 99.997% 99.997% 99.997% 99.998% 99.995% 99.994%

99.995% 99.992% 99.999% 99.993% 99.990% 99.995% 99.993% 99.999%

99.997% 99.993% 99.999% 99.996% 99.998% 99.996% 99.993% 99.995%

99.992% 99.998% 99.993% 99.993% 99.999% 99.993% 99.998% 99.996%

99.993% 99.996% 99.996% 99.994% 99.999% 99.996% 99.996% 99.992%

99.999% 99.996% 99.991% 99.996% 99.992% 99.998% 99.997% 99.994%

99.998% 99.995%

--- Results after 98 passes ---

Best: 100.000 -- Worst: 99.990 -- Average: 99.996163, Difference: 99.998235


--

 [root@asterisk ~]# cat
/sys/devices/system/clocksource/clocksource0/current_clocksource
*tsc*
 [root@asterisk ~]# cat
/sys/devices/system/clocksource/clocksource0/available_clocksource
tsc hpet acpi_pm jiffies


On 18 March 2011 17:52, Tiago Geada tiago.ge...@gmail.com wrote:

 Hi list!

 We currently have a PRI gateway composed by a box with two Digium quad-span
 PRI cards (a TE420 and a ).
 One of the cards is filled with TELCO1, while the other has first two slots
 filled with TELCO2, and 3rd slot with TELCO3.

 I am currently having (timer ?) issues on TELCO3 (span 7)

 D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing
 on-going calls to terminate.
 Problem clears immediately tho. I send a copy of the log with pri debug at
 a time of problems...

 Is there a problem having 2 telcos on the same PRI card?
 Would somebody help?

 asterisk*CLI pri show span 7
 Primary D-channel: 202
 Status: Provisioned, Up, Active
 Switchtype: EuroISDN
 Type: CPE
 Overlap Dial: 0
 Logical Channel Mapping: 0
 Timer and counter settings:
   N200: 3
   N202: 3
   K: 7
   T200: 1000
   T202: 1
   T203: 1
   T303: 4000
   T305: 3
   T308: 4000
   T309: 6000
   T313: 4000
   T-HOLD: 4000
   T-RETRIEVE: 4000
   T-RESPONSE: 4000
 Overlap Recv: No


 and

 [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (I): T200
 expired N200 times sending RR/RNR in state 8(Timer recovery)
 [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:05] VERBOSE[19844] chan_dahdi.c: Changing from state 8(Timer
 recovery) to 5(Awaiting establishment)
 [Mar 18 17:04:06] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:07] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:08] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (G): *T200
 expired N200 times sending SABME in state 5(Awaiting establishment)*
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Changing from state
 5(Awaiting establishment) to 4(TEI assigned)
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 DL event:
 Q931_DL_EVENT_DL_RELEASE_IND(3)
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=56
 on channel 2
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=64
 on channel 3
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=58
 on channel 4
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Start T309 for call cref=66
 on channel 6
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c: Changing from state 4(TEI
 assigned) to 5(Awaiting establishment)
 [Mar 18 17:04:09] VERBOSE[19844] chan_dahdi.c:   == Primary D-Channel on
 span 7 down
 [Mar 18 17:04:09] WARNING[19844] chan_dahdi.c: No D-channels available!
  Using Primary channel 202 as D-channel anyway!
 [Mar 18 17:04:10] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:11] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:12] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 MDL-ERROR (G): T200
 expired N200 times sending SABME in state 5(Awaiting establishment)
 [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: Changing from state
 5(Awaiting establishment) to 4(TEI assigned)
 [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 DL event:
 Q931_DL_EVENT_DL_RELEASE_IND(3)
 [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: TEI=0 Sending SABME
 [Mar 18 17:04:13] VERBOSE[19844] chan_dahdi.c: Changing from state 4(TEI
 assigned) to 5(Awaiting

Re: [asterisk-users] One PRI card with 2 (or more) Telcos

[Tiago Geada]

Hi! I can try that tho. Where do I configure what timer to use??!

Thanks in advance.

On 18 March 2011 18:21, Andrew Latham lath...@gmail.com wrote:

 On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini
 adrian-li...@wombit.com wrote:
 
  Is there a problem having 2 telcos on the same PRI card?
 
  I think you go with one master timer as the Telco.  Then the other spans
 are
  secondary, tertiary, quaternary timers.
 
  Adrian


 Adrian

 This only works when all the providers are using a common clock like
 some areas in the USA.  This is not the case all around the world.

 --
 ~~~ Andrew lathama Latham lath...@gmail.com ~~~

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] One PRI card with 2 (or more) Telcos

[Tiago Geada]

OK I found it.

In /etc/dahdi/system.conf

I have for this span:


# Span 7: TE4/1/3 T4XXP (PCI) Card 1 Span 3 HDB3/CCS/CRC4
span=7,7,0,ccs,hdb3,crc4
# termtype: te
bchan=187-201,203-217
dchan=202
echocanceller=mg2,187-201,203-217


should I use span=7,*5*,0,ccs,hdb3,crc4 instead? (5 is the first telco on
that card)

On 18 March 2011 18:23, Tiago Geada tiago.ge...@gmail.com wrote:

 Hi! I can try that tho. Where do I configure what timer to use??!

 Thanks in advance.

 On 18 March 2011 18:21, Andrew Latham lath...@gmail.com wrote:

 On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini
 adrian-li...@wombit.com wrote:
 
  Is there a problem having 2 telcos on the same PRI card?
 
  I think you go with one master timer as the Telco.  Then the other spans
 are
  secondary, tertiary, quaternary timers.
 
  Adrian


 Adrian

 This only works when all the providers are using a common clock like
 some areas in the USA.  This is not the case all around the world.

 --
 ~~~ Andrew lathama Latham lath...@gmail.com ~~~

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] One PRI card with 2 (or more) Telcos

[Tiago Geada]

Sorry to keep bugging, but after making changes to /etc/dahdi/system.conf,
do I need unload  res_timing_dahdi.so and chan_dahdi.so; and load them, or
can I just reload them??

Thanks in advance

On 18 March 2011 18:26, Tiago Geada tiago.ge...@gmail.com wrote:

 OK I found it.

 In /etc/dahdi/system.conf

 I have for this span:


 # Span 7: TE4/1/3 T4XXP (PCI) Card 1 Span 3 HDB3/CCS/CRC4
 span=7,7,0,ccs,hdb3,crc4
 # termtype: te
 bchan=187-201,203-217
 dchan=202
 echocanceller=mg2,187-201,203-217


 should I use span=7,*5*,0,ccs,hdb3,crc4 instead? (5 is the first telco
 on that card)

 On 18 March 2011 18:23, Tiago Geada tiago.ge...@gmail.com wrote:

 Hi! I can try that tho. Where do I configure what timer to use??!

 Thanks in advance.

 On 18 March 2011 18:21, Andrew Latham lath...@gmail.com wrote:

 On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini
 adrian-li...@wombit.com wrote:
 
  Is there a problem having 2 telcos on the same PRI card?
 
  I think you go with one master timer as the Telco.  Then the other
 spans are
  secondary, tertiary, quaternary timers.
 
  Adrian


 Adrian

 This only works when all the providers are using a common clock like
 some areas in the USA.  This is not the case all around the world.

 --
 ~~~ Andrew lathama Latham lath...@gmail.com ~~~

 --
 _
 -- 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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Re: [asterisk-users] One PRI card with 2 (or more) Telcos

[Tiago Geada]

Hi Kevin,

Thanks for your elaborated answer. I will try and set them on the same clock
and see if no problem occurs. If so, Different telco's clocks would be in
SYNC (I do doubt it).

This machine has no more PCI slots available and hardware is damn expensive.

Will have to look into it with my boss..

Thanks you.

On 18 March 2011 18:30, Kevin P. Fleming kpflem...@digium.com wrote:

 On 03/18/2011 01:23 PM, Tiago Geada wrote:

 Hi! I can try that tho. Where do I configure what timer to use??!


 If your telcos are not synchronizing their network clocks to each other,
 you will not be able to solve this problem on a multi-port Digium T1/E1
 card. Digium T1/E1 cards select a single master clock (either the onboard
 clock or the clock recovered from one of the spans) to use as the 'board
 clock', which is then used to transmit data on all the spans. If the master
 clock is not in synchronization with the clocks at the other end of those
 spans, then bit slips will occur and cause various sorts of problems. This
 is why a card is always configured to use the recovered clock from a telco
 span if there is one, because the onboard clock would never by in sync with
 it.

 If you have a board connected to two telcos and their clocks are not
 synchronized, not only will you have trouble using a Digium card, but even
 using a card that can handle using multiple transmit clocks at once will not
 solve the underlying bit slip problem that will occur if you ever connect a
 channel from Telco1 to a channel from Telco2. If you *never* connect
 channels between Telcos, then you don't have to worry about that problem,
 but if you do, at some point during the call there will be buffer overruns
 or underruns and there will be some effect (for a normal voice call, the
 effect might be a short audio artifact, and fairly harmless... unless the
 call is a modem or FAX call, in which case it could cause the call to fail).

 For your sanity, I would strongly suggest that you don't connect spans from
 multiple telcos/networks/etc. on a single card, but keep each span provider
 on their own card.

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org


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Re: [asterisk-users] Queue member status - BUSY

[Tiago Geada]

Hi.

We use GROUP and GROUP_COUNT to track if the peer is engaged in a call. If
so we use Busy()

On 22 October 2010 01:28, GBR Icasiano, Ryan A. 
raicasi...@globalbridgeresources.com wrote:

 Hi,

 I have modified the way agents are being treated since they are using
 mobile phones. Having that kind of scenario, it is not recommended to make
 the agent logged in by using that scenario. Instead, they will call a
 certain number, login by using the given parameters(company id, username,
 password) and tag them in the DB as logged in, and their number will ring
 once a client/customer calls and falls on the queue.

 Now once asterisk falls to a certain queue, it will then check all members
 that contains login status on a certain table, then add/delete them in
 queue_members table in realtime depending on its current login status. This
 way, it will only ring all currently logged in members. It works fine this
 way, the only problem is that whenever all members are engaged on a call,
 their phone is off, etc... the queue cannot determine whether any of them is
 available or not, as far as I know.

 regards,

 RYAN ICASIANO
 
 From: asterisk-users-boun...@lists.digium.com [
 asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez [
 cur...@telecomabmex.com]
 Sent: Friday, October 22, 2010 12:06 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Queue member status - BUSY

 On Thu, 2010-10-21 at 08:15 +0800, GBR Icasiano, Ryan A. wrote:
  anyone?
 
  regards,
 
  RYAN ICASIANO
 
  
  From: asterisk-users-boun...@lists.digium.com [
 asterisk-users-boun...@lists.digium.com] On Behalf Of GBR Icasiano, Ryan
 A. [raicasi...@globalbridgeresources.com]
  Sent: Wednesday, October 20, 2010 2:02 PM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] Queue member status - BUSY
 
  Hi,
 
  Is there a way to know if a member of a queue is currently engaged on a
 call? Or if a queue can return a busy status if all members are currently
 engaged in a call? QUEUESTATUS only returns FULL and TIMEOUT, and the
 scenario only falls into TIMEOUT, and has to finish the assigned number of
 seconds into the QUEUE CMD before it falls back to the next routine on the
 dialplan.
 
  Any ideas?
 
People do not really get that a queue is supposed to work that way.
 The point of having a queue is that you will have more people waiting
 than agents available to answer calls, if not why have a queue just make
 a dial group.

The way to do what you want would be to use an AGI that gets a list
 of
 agents logged into the queue and see their status.  The status for a
 free agent is 1 so if you do not see any agents with status 1 then all
 agents are busy.  You can then set a variable so you can redirect the
 caller somewhere else.

 --
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 Carlos Chávez Prats
 Director de Tecnología
 +52-55-91169161 ext 2001

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Re: [asterisk-users] How to find .gsm audio file length or duration

[Tiago Geada]

r you would have to convert that gsm to another format first like ogg

On 16 October 2010 18:23, Barry Miller asterisk-us...@notanet.net wrote:

 On Sat, Oct 16, 2010 at 04:12:14PM +0530, RAJNIKANT VANZA wrote:
  Hi Friends,
 
  I need to find .gsm file length or duration.
 
  *E.g.*
  demo-congrats.gsm
 
  sox demo-congrats.gsm -e stat
 
  Above command is display file length in seconds. like as
  Length (seconds): 27.96
 
  I want to .gsm file length or duration in dialplan.

Set(DUR=$[${STAT(s,/var/lib/asterisk/sounds/en/demo-congrats.gsm)} /
 1650])
   Verbose(Length (seconds): ${DUR})

 for asterisk = 1.6

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Re: [asterisk-users] What can make G.729a codec hostid change?

[Tiago Geada]

Hi,

I don't have any g729 codec license. But by reading Barry's complaint I get
to think that it is really unfair that Digium can't renew his license or
something.

I am a Debian user myself and I understand the need to upgrade from etch to
lenny (and to squeeze in no time).
Having a kernel built on purpose to remove some modules is out of line.

A better solution needs to be provided in cases like these.

On 7 September 2010 19:15, Roger Burton West ro...@firedrake.org wrote:

 On Tue, Sep 07, 2010 at 10:58:18AM -0700, Dave Platt wrote:

 Note that ifconfig will not necessarily show all of your
 interfaces (hard- or soft-) - only the active, configured ones.

 ifconfig -a would help here. Kernel upgrades often seem to bring in new
 default interfaces.

 If this turns out to be the problem, rmmod or a custom kernel
 compilation may do the trick. (Of course if you've _lost_ an interface
 you were using under etch this may be more of a problem.)

 R

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Re: [asterisk-users] CDR variables

[Tiago Geada]

Ow...

I have =no commented, so I guess =yes is default??
; Normally, CDR's are not closed out until after all extensions are finished
; executing.  By enabling this option, the CDR will be ended before
executing
; the h extension so that CDR values such as end and billsec may be
; retrieved inside of of this extension.
;endbeforehexten=no

So if I uncomment that, I will be able to use billsec in h exten... right?

Thanks Danny!

On 18 August 2010 22:19, Danny Nicholas da...@debsinc.com wrote:

   *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tiago Geada
 *Subject:* [asterisk-users] CDR variables



 Hello list!

 I am trying to get hold of ${CDR(duration)} and ${CDR(billsec)} variables
 in h

 It seems that these variables always return 0. I am using  Asterisk
 version 1.6.2.11. Can't I get these values other than using CDR reccords ??



 In cdr.conf, is endbeforehexten=yes ?

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[asterisk-users] CDR variables

[Tiago Geada]

Hello list!

I am trying to get hold of ${CDR(duration)} and ${CDR(billsec)} variables in
h

It seems that these variables always return 0. I am using  Asterisk version
1.6.2.11. Can't I get these values other than using CDR reccords ??
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Re: [asterisk-users] sending sms from Asterisk server

[Tiago Geada]

I would rather use .call files. So easy to produce a text file...

On 18 August 2010 21:02, Steve Edwards asterisk@sedwards.com wrote:

 Un-top-posting...

  On 08/17/2010 09:00 AM, Tino wrote:

 I would like to send sms to some external phone numbers from my asterisk
 server. Is it possible to send sms via softphones like X-Lite ? . Any tips
 regarding this will be helpful


  On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn johann.ho...@ecommerce.com
 wrote:


  This is easy to do by using email to SMS gateways.  A list of them is on
 wikipedia (http://en.wikipedia.org/wiki/List_of_SMS_gateways).  For the
 Asterisk side, you have an extension that sends the email.  I personally use
 an AGI script for this part, but you could use a System() call as well.


 Using system() is almost always a hack -- and not the good kind :)


 On Wed, 18 Aug 2010, Tino wrote:

  Thanks for your advice in this matter. But i am not sure how to pass the
 numbers to be sent sms  in the dialplan.


 You have a choice: you can pass them as channel variables or as command
 line options. I use both, frequently in the same program. Unfortunately, I
 can't clearly articulate why I use one over the other. If the variable is
 something that exists for the life of the call like ${CLIENT-ID} I tend to
 access it as a channel variable. If it's something that modifies the
 behavior of the AGI (--debug or --verbose) I always pass it as a command
 line option and use getopt_long()

 First, you need to pick a language. If this is a SOHOish hobby project, it
 doesn't matter -- pick a language you are comfortable with.

 If this is a high volume, performance critical project -- I'd vote for c.

 Once you've decided on a language, search out an established AGI library
 and learn a bit about the protocol. It's very simple but not always obvious.
 The 3 biggest stumbling blocks that trip up programmers are:

 1) You have to read the AGI environment before anything else.

 2) It's a request followed by a response. If you don't read the response,
 bad things will happen.

 3) It's STDIN/STDOUT based. If you try to debug by writing variables or
 messages using echo/printf/puts/etc, bad things will happen.

 Check out voip-info.org for more information on AGI.

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000
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Re: [asterisk-users] realtime sip peers : musiconhold class

[Tiago Geada]

Hi.

Just to let you know, we record voices with audacity, and export audio as
flac, just in case we need to edit it.

Then I have the following sh script:

o# cat convert.sh
#!/bin/sh

today=$(date +%F);

mkdir -p $today/flac;
mkdir -p $today/wav;
mkdir -p $today/ul;

for i in *.flac;
do
echo 
echo Processing $i;
echo 
#$filename=
sox $i -r 8000 -c 1 $(echo $i|rev|cut -d . -f2-10|rev).wav;
normalize-audio -a 25dB $(echo $i|rev|cut -d . -f2-10|rev).wav;
mv $i $today/flac/;
sox $(echo $i|cut -d . -f1).wav $(echo $i|rev|cut -d .
-f2-10|rev).ul;
mv $(echo $i|rev|cut -d . -f2-10|rev).wav $today/wav/;
mv $(echo $i|rev|cut -d . -f2-10|rev).ul $today/ul/;
echo ;
done

echo All done;


On 17 August 2010 08:07, Jonas Kellens jonas.kell...@telenet.be wrote:

  Can anyone help because I don't understand why Asterisk can not read the
 input file, there is not much info given...

 2 files :

 [r...@asterisk testing]# file testExtended.wav
 testExtended.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16
 bit, stereo 44100 Hz
 [r...@asterisk testing]# file testLong.wav
 testLong.wav: RIFF (little-endian) data, WAVE audio, 20294 channels
 1414676809 Hz

 to mono :

 [r...@asterisk testing]# sox testExtended.wav -r 8000 -c1
 testExtended2.wav resample -ql

 sox sox: effect `resample' is deprecated; see sox(1) for an alternative
 [r...@asterisk testing]# sox testLong.wav -r 8000 -c1 testLong2.wav
 resample -ql

 sox sox: effect `resample' is deprecated; see sox(1) for an alternative
 sox effects: resample clipped 2 samples; decrease volume?

 afterwards :

 [r...@asterisk testing]# file testLong2.wav
 testLong2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16
 bit, mono 8000 Hz
 [r...@asterisk testing]# file testExtended2.wav
 testExtended2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16
 bit, mono 8000 Hz

 But Asterisk can not open them :

 [r...@asterisk testing]# asterisk -rx file convert testExtended2.wav
 testExtended2.alaw
 Unable to open input file: testExtended2.wav
 [r...@asterisk testing]# asterisk -rx file convert testLong2.wav
 testLong2.alaw
 Unable to open input file: testLong2.wav


 Any thoughts ?!


 Jonas.



 On 08/14/2010 04:30 PM, Motiejus Jakštys wrote:

 On Sat, Aug 14, 2010 at 4:24 PM, Jonas Kellens jonas.kell...@telenet.be
 wrote:
 
  intro extended version.wav: RIFF (little-endian) data, WAVE audio,
 Microsoft
  PCM, 16 bit, stereo 44100 Hz
 

 You need *MONO, 8000Hz*

 $ man sox

 --
 Motiejus Jakštys


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Re: [asterisk-users] What‘s the best operating syst em suggest for Asterisk 1.6.2.9

[Tiago Geada]

That would be probably because Ubuntu became top-famous and widely used for
anything, just fashion so to speak, while CentOS is probably chosen because
asterisknow runs on top of centos.

On 30 June 2010 12:30, Leif Madsen leif.mad...@asteriskdocs.org wrote:

 I'm not entirely sure I see where he implied it was. His answer refers to
 the
 question, I want to know what is the best OS for installing Asterisk...?

 I like both CentOS and Ubuntu. The next edition of the O'Reilly Asterisk
 book
 will cover installing Asterisk on both OS's.

 Leif.

 Tiago Geada wrote:
  Ubuntu is not Debian.
 
  I would recommend Debian tho, its rock solid and it jsut works (for
  anything)
 
  On 29 June 2010 12:29, Paul Belanger paul.belan...@polybeacon.com
  mailto:paul.belan...@polybeacon.com wrote:
 
  On Mon, Jun 28, 2010 at 10:04 PM, Zhang Shukun bit...@gmail.com
  mailto:bit...@gmail.com wrote:
i want to know what is the best OS for install Asterisk
 1.6.2.9,
which should work properly on working system.
   
  Ubuntu 10.04 Server ?

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Re: [asterisk-users] What‘s the best operating syst em suggest for Asterisk 1.6.2.9

[Tiago Geada]

Ubuntu is not Debian.

I would recommend Debian tho, its rock solid and it jsut works (for
anything)

On 29 June 2010 12:29, Paul Belanger paul.belan...@polybeacon.com wrote:

 On Mon, Jun 28, 2010 at 10:04 PM, Zhang Shukun bit...@gmail.com wrote:
  i want to know what is the best OS for install Asterisk 1.6.2.9,
  which should work properly on working system.
 
 Ubuntu 10.04 Server ?

 --
 Paul Belanger | dCAP
 Polybeacon | Consultant
 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 blog.polybeacon.com

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Re: [asterisk-users] NO ANSWER before playback or background function?

[Tiago Geada]

We use a dial option A() that will stream audio as soon as the calle picks
up...

On 23 June 2010 05:50, Zhang Shukun bit...@gmail.com wrote:

 2010/6/22 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de:
  Hi!
 
  but i want to answer the channel when dial someone and pick up the
  phone.not play a file.
 
  Search this list for early media and maybe also for progress.

 Thanks , i have search for early media, and have get some valuable
 infomation.

 i can play files with noanswer .

 exten = s,1,Progress
 exten = s,n,Playback(hello,noanswer)  ;this works good.
 exten = s,n,Dial(SIP/1...@bd-test,30)
 exten = s,n,Playback(hello,noanswer) ; this works no sound

 the first Playback works good. i can hear the sound and it won't
 answer the channel first.

 my problem is after Dial command, if not answer the channel(connected).

 next will execute: exten = s,n,Playback(hello,noanswer) ; this works no
 sound

 but this Playback have no sound.

 Do you know what's wrong?



 
  Philipp
 


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Re: [asterisk-users] Asterisk distribution for a Call Center

[Tiago Geada]

Plain asterisk. You only configure it once, and re-use the configuration for
different call centers :-)


On 23 June 2010 00:28, Luciano Moreira lmore...@dxbrasil.net wrote:

 We use Vicidial for all size CallCenter. It's very powerful for multi
 server and/or multi site. We have vicidial from tiny callcenter one
 site with 5 agents to over 1000 Agents distributed in 20 cities
 working as just one callcenter.

 Info http://astguiclient.sourceforge.net/vicidial.html

 __
 Luciano Moreira

 Logic Telecom LTDa
 Fortaleza, CE

 +55 (85) 4062-9150
 +55 (85) 9701-2444
 +1 360-717-1506 (USA)



 2010/6/22 Tarek Sawah tareksa...@hotmail.com:
  i have been struggling with call center Customers for a couple of years
  now.. i have a call center with 40 agents using elastix.. and quality is
  related to the source of calls inbound or outbound...
  the problem with call centers they need Visual .. like Flash Operator
 panel
  and CDRs..
  if you can go with simply raw asterisk .. without any additions.. will be
  the best for you .. write your own dial plans.
  Flash operator Panel is not a flawless work.. and adds more burden on the
  resources.. esp when it's open by 7-8 persons at once..
  regarding the ACD ..it's all about PHP and Database .. you can build your
  own reports and so. or you can use a2billing to do the billing and ACD..
  Elastix has a good billing (without a2billing) .. but i prefer a clean
  installation of asterisk and work around with database and PHP much
  better..
  Good Luck!
 
  -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1
  (386) 492-9993
 
 
  Date: Tue, 22 Jun 2010 15:21:18 -0300
  From: aco1...@gmail.com
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] Asterisk distribution for a Call Center
 
  Dear all, I need to build a PBX based on Asterisk for a call center. I
  have worked with raw Asterisk but it's hard to work for big
  implementations think.
 
  Also I have worked with Trixbox CE for a small bussines and it was
  prette good, but I have not have many features like ACD. I know there
  is another version called Trixbox PRO -specially Call Center edition-
  that's not free but has got more features like ACD and billing.
 
  I've heart about AsteriskNow and I know it's free.
 
  What distribution/version do you recommend to me in order to implement
  a call center and taking into account I'm not an expert in programming
  from Asterisk CLI ???
 
  Thanks a lot
 
  Alejandro
 
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Re: [asterisk-users] realtime queues membername problem

[Tiago Geada]

to re-read peers from realtime db try: sip prune realtime all

On 23 June 2010 01:22, Jean Chassoul chass...@gmail.com wrote:

 anyone know something about this?


 On Fri, May 14, 2010 at 10:56 AM, Jean Chassoul chass...@gmail.comwrote:

 Hi,

 I'm using dynamic realtime with asterisk 1.6.0.24, I'm having a strange
 problem with queue_members...

 If I update only 'membername' field on queue_members table asterisk won't
 refresh the change, but if I update another field like interface everything
 works as expected, I've found this problem also deleting a existing agent on
 queue_members and then inserting a new one with the same interface, penalty
 and pause but with another membername :( Asterisk won't refresh the change
 and show the old membername on CLI  (queue show my-queue...).

 It is possible that asterisk refresh these info?

 Thanks.




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Re: [asterisk-users] Local channel usage

[Tiago Geada]

Hi,

After a Dial, the call is hung up. It doesn't carry on with dialplan unless
you specify the appropriate dial option.

Check wiki voip-info for cmd Dial, I think the option is g

2010/6/22 Harel Cohen ha...@easycall.gi

 Hi All,

 I’m trying to do “things” after my Dial application terminates (e.g. play
 IVR to called party, calling party, etc.). I’m trying to use the local
 channel for this purpose but so far with no success. I’m using 1.6.1.18 and
 this is my extensions.conf:



 [Internal]

 exten = _22,1,Dial(Local/${ext...@cw/n) ; 22 is test number

 exten = _22,2,Noop(After Hangup)



 [CW]

 exten = _x.,1,Dial(SIP/307)

 exten = _x.,2,Noop(After Hangup)



 The call never reaches neither of the Noop applications. Consol:

   == Using SIP RTP CoS mark 5

   == Using UDPTL CoS mark 5

 -- Executing [...@internal:1] Dial(SIP/309-00a5, Local/2...@cw/n)
 in new stack

 -- Called 2...@cw/n

 -- Executing [...@cw:1] Dial(Local/2...@cw-af6f;2, SIP/307) in new
 stack

   == Using SIP RTP CoS mark 5

   == Using UDPTL CoS mark 5

 -- Called 307

 -- SIP/307-00a6 is ringing

 -- Local/2...@cw-af6f;1 is ringing

 -- SIP/307-00a6 is ringing

 -- SIP/307-00a6 is ringing

 -- SIP/307-00a6 is ringing

 -- SIP/307-00a6 answered Local/2...@cw-af6f;2

 -- Local/2...@cw-af6f;1 answered SIP/309-00a5

   == Spawn extension (CW, 22, 1) exited non-zero on 'Local/2...@cw-af6f;2'

   == Spawn extension (Internal, 22, 1) exited non-zero on
 'SIP/309-00a5'

 If I use the ‘g’ option in my Dial() both Noop will be run only if the
 called party hang-up first. I need a simple continuation in the dial plan
 regardless of who disconnected the call.

 Thanks in advance

 Harel



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Re: [asterisk-users] Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

[Tiago Geada]

Hi!

If it was me, I would create a bash script calling asterisk -vrx core show
commands

something like:

for chan in $(asterisk -vrx core show channels concise);
do
asterisk -vrx core show channel $(echo $chan|cut -d \! -f1)|grep -i
native;
done

On 21 June 2010 16:08, bruce bruce bruceb...@gmail.com wrote:

 Hi Everyone,

 I want to know if a specific codec type is used at least one. For example,
 I want to know if out of the 100 calls on the system if there is a 1 channel
 that is running G.729 codec right now. If using dial-plan and I dial in, I
 can use this to obtain information about CURRENT channel. But it won't allow
 me to obtain information about OTHER channels and that is what I want to do.
 I want a search for all channels and an output spit out as g729 or TRUE or
 FALSE if there is a g729 channel.

 exten = s,1,Answer()
 exten = s,n,Set(foo=${CHANNEL(audioreadformat)})
 exten = s,n,NoOp(${foo})

 Above  NoOp spits out g729 if I call in with a g729 codec. But I want 
 that to be about other channels and not the one I am calling into.

 Thanks,

 Bruce


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[asterisk-users] SMS in landline

[Tiago Geada]

Hi all.

I am searching for a way to send SMS via our E1 PRI line.

We are in Portugal and I have seen some Internet/TV/Phone providers (ZON for
those who know it) who install normal phones with SMS support in landline.

So I just found a page from PT (Portugal Telecom) stating that the SMC
number is either 12999 or 129990 (
http://www.ptcom.pt/PTResidencial2/Tabs/MyPTPublico/Apoio_a_Clientes/Servi%C3%A7os/SMS/caracteristicas/sms_caracteristicas.htm
)

Now I was trying to send a SMS via a PRI from PT (same provider)

context of dialplan is services

[r...@asterisk ~]# tail /etc/asterisk/extensions_services.ael -n 12
_00019 = { // TEST SMS
Noop(Testing SMS to ${EXTEN:4}...);
Answer();
SMS(services,,00351932485457,bla);
SMS(services);
Hangup();
//  129990
}

/ FINISHED TESTING /

}
 [r...@asterisk ~]# cat test.call
Channel: DAHDI/g7/12999
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: services
Extension: 0001932485457
Priority: 1
SetVar: MSG=hello


cp test.call /var/spool/asterisk/outgoing/  chown asterisk.asterisk
/var/spool/asterisk/outgoing/test.call  chmod 777
/var/spool/asterisk/outgoing/test.call  asterisk -vvr

Asterisk 1.6.2.9-rc2, Copyright (C) 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty'
for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
  == Parsing '/etc/asterisk/asterisk.conf':   == Found
  == Parsing '/etc/asterisk/extconfig.conf':   == Found
Connected to Asterisk 1.6.2.9-rc2 currently running on asterisk (pid = 12521)
Verbosity is at least 14
-- Attempting call on DAHDI/g7/12999 for 0001932485...@services:1 (Retry 1)
-- Making new call for cr 32792
-- Requested transfer capability: 0x00 - SPEECH
 Protocol Discriminator: Q.931 (8)  len=28
 Call Ref: len= 2 (reference 24/0x18) (Originator)
 Message type: SETUP (5)
 [04 03 80 90 a3]
 Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: 
 Speech (0)
  Ext: 1  Trans mode/rate: 64kbps, circuit-mode 
 (16)
User information layer 1: A-Law (35)
 [18 03 a1 83 81]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0  Preferred  
 Dchan: 0
ChanSel: As indicated in following octets
   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
   Ext: 1  Channel: 1 ]
 [6c 02 21 80]
 Calling Number (len= 4) [ Ext: 0  TON: National Number (2)  NPI: 
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)
   Presentation: Presentation permitted, user number 
 not screened (0)  '' ]
 [70 06 a1 31 32 39 39 39]
 Called Number (len= 8) [ Ext: 1  TON: National Number (2)  NPI: 
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)  '12999' ]
 [a1]
 Sending Complete (len= 1)
q931.c:3134 q931_setup: call 32792 on channel 1 enters state 1 (Call Initiated)
 Protocol Discriminator: Q.931 (8)  len=32
 Call Ref: len= 2 (reference 24/0x18) (Terminator)
 Message type: CALL PROCEEDING (2)
 [18 03 a9 83 81]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit  PRI  Spare: 0
Exclusive  Dchan: 0
ChanSel: As indicated in following octets
   Ext: 1  Coding: 0  Number Specified  Channel Type: 3
   Ext: 1  Channel: 1 ]
 [28 14 43 48 41 4d 41 44 41 20 45 4d 20 50 52 4f 47 52 45 53 53 4f]
 Display (len=20) [ CHAMADA EM PROGRESSO ]
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 40 (cs0, Display)
q931.c:3683 q931_receive: call 32792 on channel 1 enters state 3
(Outgoing call  Proceeding)
 Protocol Discriminator: Q.931 (8)  len=52
 Call Ref: len= 2 (reference 24/0x18) (Terminator)
 Message type: DISCONNECT (69)
 [08 02 84 9c]
 Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0
Location: Public network serving the remote user (4)
  Ext: 1  Cause: Invalid number format (28), class =
Normal Event (1) ]
 [1c 17 91 a1 14 02 01 2e 02 01 24 30 0c 30 0a a1 05 30 03 02 01 00 82 01 00]
 Facility (len=25, codeset=0) [ 0x91, 0xA1, 0x14, 0x02, 0x01, '.',
0x02, 0x01, '$0', 0x0C, '0', 0x0A, 0xA1, 0x05, '0', 0x03, 0x02, 0x01,
0x00, 0x82, 0x01, 0x00 ]
PROTOCOL 11
A1 0014 (CONTEXT SPECIFIC [1])
  02 0001 2E (INTEGER: 46)
  02 0001 24 (INTEGER: 36)
  30 000C (SEQUENCE)
30 000A (SEQUENCE)
  A1 0005 (CONTEXT SPECIFIC [1])
30 0003 (SEQUENCE)
  02 0001 00 (INTEGER: 0)
  82 0001 00 (CONTEXT SPECIFIC [2])
 [1e 02 82 88]
 Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT 

Re: [asterisk-users] own Caller ID

[Tiago Geada]

We can set our own CallerID. Telco gives us 100 different numbers comming in
our PRI and we may choose one of those 100 as a CallerID

We had to ask telco to permit us this change.

They allowed us to set on the initial SETUP message if we use our own
presentation.

This we we can also use Callerpresentation = prohib

also set this directive on chan_dahdi.conf:
usecallingpres=yes

On 8 June 2010 20:44, Steve Edwards asterisk@sedwards.com wrote:

 On Tue, 8 Jun 2010, taimur hasan wrote:

  I want to use my own caller id, instead of the caller id of PSTN line,
 for the outbound calls through DAHDI channel. Is there any way ??


 It depends on your technology (POTS, PRI, etc) and your provider.

 Tell your provider you want to set the outgoing caller ID and see what
 their response is.

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000
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Re: [asterisk-users] problem with inserting records into cdr

[Tiago Geada]

pasting the error would help

On 3 June 2010 20:56, cov...@ccs.covici.com wrote:

 Hi.  For several months now asterisk will mysteriously stop inserting
 records into cdr database.  I am using mysql and the asterisk addons
 1.6.2 to accomplish this.  Sometimes there is a strange error about
 column names, but often there is no error, it just stops.  I just have
 to restart asterisk to get things going again, so I am stumped as to
 what is happening, or even how to troubleshoot.  I usually run in
 verbosity 4, but am not seeing anything of interest.

 Any ideas would be appreciated.

 --
 Your life is like a penny.  You're going to lose it.  The question is:
 How do
 you spend it?

 John Covici
 cov...@ccs.covici.com

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Re: [asterisk-users] Asterisk on Ubuntu

[Tiago Geada]

either create a init script or place a crontab entrey @reboot

On 4 June 2010 13:40, Danny Dias ing.diasda...@gmail.com wrote:

 Hello Asterisk users,

 I'm having a little problem with an Asterisk installation on Ubuntu, i had
 installed many asterisks on CentOS but never in Ubuntu, the problem is that
 Asterisk and DAHDI does not start at system start...i have to make
 /etc/init.d/asterisk start and /etc/init.d/dahdi start manually every
 time i reboot the machine (my laptop for testing)

 So, what should i do in order to solve this situation?

 Thanks in advance

 Regards

 --
 Saludos
 Danny Dias
 SkypeID: danny.dias1

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Re: [asterisk-users] Connect mobile to asterisk

[Tiago Geada]

Hi,

In the past I had a sony ericsson connected via usb cable, and used gnokii
to interact with it. Then I would use System() in asterisk to call up
gnokii.

I never tried it but I think asterisk-addons has got a module to use
Bluetooth mobile phones.

On 29 May 2010 07:01, Nivin Kumar nivinkuma...@yahoo.in wrote:

 Guys,

 I would like to connect my blackberry or any other cell phone to asterisk
 so that I can send calls through the sim card. I would also like to send SMS
 through this as well. Could someone point me in the right direction?

 Thanks,
 Nivin


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Re: [asterisk-users] FAX 2 mail configuration

[Tiago Geada]

Hi João.

We made up a script that sends received faxes trough a smtp server as an
attachment.

the FAX.ael

context FAX

{

s = {

Answer();

Set(TIMEOUT(absolute)=600); // 10 min

Wait(3);

if(${CALLERID(num)}=) { //

Set(Number=withhold);   // If number
is private

}   //

else {

Set(Number=${CALLERID(num)});   // If number
is NOT private

}

Set(recordFile=${UNIQUEID}_${Number}.tiff);
// Record file to RAM first,


 
Set(recordPath=/var/log/asterisk/fax/${CALLERID(dnid)}/${STRFTIME(${EPOCH},GMT+0,%F)});
// then run /usr/local/bin/mailfax $1 $2

ReceiveFax(/ramdrive/${recordFile});

Wait(5);

Hangup();

};

h = {



System(/usr/local/bin/faxmail ${recordPath}
${recordFile});

};

}



and the script @ /usr/local/bin/faxmail has got something like:


#!/bin/sh

PATH=/usr/sbin:/sbin:/bin:/usr/bin:/usr/local/bin


if [ -d $1 ]; then

mv /ramdrive/$2 $1;

chmod a+rx $1/$2;

else

mkdir -p $1;

mv /ramdrive/$2 $1;

chmod a+rx $1/$2;

fi


#chmod a+rx /ramdrive/$2;


{

  (

sleep 1

echo ehlo tretas.eu

sleep 1

echo AUTH LOGIN

sleep 0

echo -n aster...@tretas.eu|base64

sleep 0

echo -n tretas|base64

echo MAIL FROM: aster...@tretas.eu

sleep 0

echo RCPT TO: tiago.ge...@gmail.com

echo RCPT TO: f...@tretas.eu

sleep 1

echo data


echo Subject: FAX $2

echo FROM: aster...@tretas.eu

echo TO: f...@tretas.eu

sleep 1

echo 'Content-Type: multipart/mixed; boundary=Y3VzY28udHJldGFzLmV1'

echo 


echo --Y3VzY28udHJldGFzLmV1

echo 'Content-Type: multipart/alternative;
boundary=Y3VzY28udHJldGFzLmV2'

echo 


echo --Y3VzY28udHJldGFzLmV2

echo 'Content-Type: text/plain; charset=ISO-8859-1'

echo 


echo Fax em $(date)

echo $1/$2

echo 


echo --Y3VzY28udHJldGFzLmV2

echo 'Content-Type: text/html; charset=ISO-8859-1'

echo 

echo Fax em $(date)br$1/$2

echo 


echo --Y3VzY28udHJldGFzLmV2--

echo --Y3VzY28udHJldGFzLmV1

echo 'Content-Type: image/tiff; name=fax.tiff'

echo 'Content-Disposition: attachment; filename=fax.tiff'

echo Content-Transfer-Encoding: base64

echo X-Attachment-Id: 0.1

echo 

sleep 1;


cat $1/$2|base64

sleep 1;


echo --Y3VzY28udHJldGFzLmV1--


echo .

#echo quit

   ) | telnet smtp.tretas.eu 25

}


Boa sorte!


On 30 March 2010 16:29, Joao Gomes Pereira gomespere...@startel.pt wrote:

 Hello
 Im trying to configure Fax2Mail in my Asterisk 1.4.23.1 server, wich
 receievs the Faxes through a SIP trunk.
 I found a lot of solutions in voip-info.org
 So, I would like to know what's the best free Fax2Mail solution and if I
 really need to install Dahdi or Zaptel.
 Thanks
 Regards
 Joao Pereira

 --
 StarTel - A Rede Livre
 Joao Gomes Pereira
 www.startel.pt
 +351 304500650
 sip: gomespere...@startel.pt


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Re: [asterisk-users] Asterisk Manager Interface (AMI) proxy recommendation

[Tiago Geada]

Hi!

You can just add several users to manager.conf or you can use AstManProxy...

On 21 March 2010 20:27, Leo Burd l...@media.mit.edu wrote:

 Hello there,

 I'm new to Asterisk and I'm trying to figure out a way to make the
 Asterisk Manager Interface (AMI) accessible to multiple users at the
 same time.  Would anyone recommend an AMI proxy that could be accessed
 from PHP code?

 Thanks in advance,

 Leo





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Re: [asterisk-users] Really Silly Question From Total Newbie

[Tiago Geada]

Hello there!

If your box has a live Internet connection, then all you need is a sip
provider.

Back to when I lived in the UK, there was this voipuser.org which gave me
a fixed british number for free, and some outbound call minutes too.

I'm sure that if you search around for SIP Providers, you may be able to
find some free stuff.
I believe that Outbound calls cost money, not incoming calls. I'm not
totally sure tho.

Anyway, you should find a provider and try to register with them,


-

Regards,

Tiago Lourenço Geada

2010/1/5 UIT DEVELOPMENT uit...@gmail.com

 Jamie - I will check that out!  Thanks!   It is just for testing and
 yes, the Asterisk box is connected to the Internet.  Cool.

 -M

 On Tue, Jan 5, 2010 at 4:39 PM, Jamie A. Stapleton
 jstaple...@computer-business.com wrote:
  Could use the free http://www.sipgate.com/one for some testing (assuming
 that Asterisk is connected to the Internet)
 
  -Original Message-
  From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of UIT DEVELOPMENT
  Sent: Tuesday, January 05, 2010 2:54 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [asterisk-users] Really Silly Question From Total Newbie
 
  Hello All -
 
  I've been poking around the past few weeks, trying to familiarize
  myself with all of this.  I am new to Linux, VoIP and Asterisk -- to
  be complete.   This is my first exposure to all of these technologies.
 
  I installed AsteriskNow on my old dual Pentium 833mhz Dell PowerEdge
  2400 and the install went well.   I can log in and poke around in
  Linux and I even configured the box to be recognized on my windows
  network.  However, is there a GUI that I can access to help me set
  things up?  I've gotten so far as what looks to me like DOS windows
  that I can change various things in the OS...
 
  I do not have any other hardware installed.  No cards and no VoIP
  phones.   I havent got to the point where I can make a test call or
  anything like that.  I dont know how to tell if Asterisk is up and
  running and how I can tweak it, etc.   I've been reading a lot of
  different things, and have become a bit confused. I think that in time
  it will come to me but I needed to stop and ask because I need to know
  if I am on the wrong path for what I'd like to do someday
 
  My main question is: CAN I make call from that box to my cell phone
  using a soft-phone?   If so, how can I do that?   Also, can I use my
  cell phone to call into that box?   I dont know if I have to get a
  phone number, or do I NEED a phone number?   At the moment, I do not
  have any dollars to throw at this project.   Its purely for learning,
  proof of concept sort of thing for myself on my spare time in the
  evenings.  I would simply like to be able to call out and be able to
  call into that box.  Later on down the road maybe I will get into
  setting up an IVR using a database so I can call into that system from
  wherever and get information read back to me.  But, first things
  first  I'd like to know if I am heading down the wrong path here.
 
  Sorry for what might seem as really silly questions, but I am not sure
  how to proceed.
 
  Thanks in advance for any insight that you folks can provide!
 
  Mike
 
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[asterisk-users] DEVICE STATE In use

[Tiago Geada]

Hi

We have an operator that his device state on all queues is In use where it
should be Not in use.

how can we manually change the state of a device?

I looked into the devstate function and tryed the following:

perfpbxr*CLI devstate list
perfpbxr*CLI
-
--- Custom Device States 
-
---
--- Name: 'Custom:notinuse'  State: 'NOT_INUSE'
---
-
-

then tried:

[default]

exten = ,hint,Custom:notinuse

So when any body dials  would change the devstate back to NOT_INUSE

doesn't seem to work.

How can we set the devstate?
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[asterisk-users] Script to show asterisk stuff

[Tiago Geada]

Hello folks.

I'm looking into having a web page displaying asterisk callers.
We are a call centre, and having operators answering calls at home or
whatever, they would need to have a real time application to display how
manny callers are queuing, for how long etc.

At first, I thought of phpagi. It connects to the manager and does a core
show channels concise.
This has most of the info I want.
After tweaking with php to parse the text to exatcly how I wanted, I found
out that the script would be slow if it was self refreshing (say 2 secs) and
with about 30 people opening it at the same time.

So now I was thinking in a script that would connect to the Manager, and
parse that output into a mysql table.
A Web page would consult the mysql table, showing the wanted results.

Then I thought twice and maybe some of you already developed a situation
like this and would not mind sharing?

I don't mind sharing the little I done so far, if anyone is interested.


Thanks all
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Re: [asterisk-users] Script to show asterisk stuff

[Tiago Geada]

2010/1/4 Will Payne w...@teambadger.co.uk


 On 4 Jan 2010, at 16:46, Tiago Geada wrote:

  Hello folks.
 
  I'm looking into having a web page displaying asterisk callers.
  We are a call centre, and having operators answering calls at home or
 whatever, they would need to have a real time application to display how
 manny callers are queuing, for how long etc.
 
  At first, I thought of phpagi. It connects to the manager and does a
 core show channels concise.
  This has most of the info I want.
  After tweaking with php to parse the text to exatcly how I wanted, I
 found out that the script would be slow if it was self refreshing (say 2
 secs) and with about 30 people opening it at the same time.
 
  So now I was thinking in a script that would connect to the Manager, and
 parse that output into a mysql table.
  A Web page would consult the mysql table, showing the wanted results.

 Or, if you want less work..  have a script which connects to the manager,
 formats the data and creates an HTML page. Then wait x seconds and loop.

 Then, home workers just view that one static page and use a meta-refresh or
 something.. Only one script is doing any real work and serving a static page
 to clients shouldn't overload the server.


 Will
 __


Hi Will.

Thanks for replying.

That was sort of my second thought. But once I connect to the manager I can
listen to all the events, Calls comming in, which extension they are dialed
to, lots of info... so I just got sort of confused for whitch path I should
take.

I guess I will do just that.

Thanks

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Re: [asterisk-users] monitor-type=MixMonitor

[Tiago Geada]

yes, sox is installed.

Anyway, I changed the lines that read: Monitor(gsm,/var/log) to
MixMonitor(/var/log/file.gsm...)

Thanks for answering.

2009/12/16 Holger von Ameln holger.von.am...@peercom.net

 This may be pretty obvious but do you have sox installed? I managed to
 forget that on more than one occasion ;-)

 --
 Holger von Ameln
 Peercom Ltd.  Co. KG

 
 holger.von.am...@peercom.net
 Tel.: +49 (0) 511-84887106
 http://www.peercom.net/peercomshop
 
 GF: Kati von Ameln
 Weiße Hube 2a
 D-30519 Hannover
 
 USt.-IdNr.: DE262241650
 

 Absenderkennzeichnung gem. §37a HGB, §80 Abs.1 S.1 AktienG sowie §35a Abs.1
 S.1 GmbHG:

 Peercom Ltd.  Co. KG, eingetragen im Handelsregister Hannover unter HRA
 201164 Geschäftsführung Kati von Ameln, Sitz der Gesellschaft ist Hannover.



 Am 15.12.2009 um 19:41 schrieb Tiago Geada:

  Hi!
 
  Since we upgraded to 1.6.1.11, asterisk is only outputing monitored files
  -in and -out.
  It is not mixing them in the end.
 
  queues.conf has monitor-type=MixMonitor...
 





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[asterisk-users] member (In use)

[Tiago Geada]

Hello list.

We just upgraded to 1.6.1.11.

We are using real time information stored on mysql databases. That is all
running fine.

Now, since we upgraded, some member don't get calls from queues.
In CLI: queue show shows something like:
611 (Local/6...@agents) with penalty 20 (realtime) (*In use*) has taken no
calls yet


We use the extension 611 in different computers, in the internal network
with no nat, in the external network with nat...
We deleted the member 611 from mysql, and added it again, changed passwd
etc...
We restarted asterisk several times..

The member shows always (In use) !!

Just to show that there is no channel associated with the member
core show channels shows:
Connected to Asterisk 1.6.1.11 currently running on perfpbxr (pid = 12955)
Channel  Location State
Application(Data)
DAHDI/9-1m...@fnacsaclojas:2  Up
Playback(audio/FnacSAC/qualida
DAHDI/31-1   s...@zon:7  Up
BackGround(audio/ZON/prima1)
SIP/209-0570 m...@agents:1Up  AppQueue((Outgoing
Line))
SIP/604-056e t...@agents:1   Up  AppQueue((Outgoing
Line))
DAHDI/5-1m...@fnacsacbilhetei Up
Queue(FnacSACBilheteira,t,,,18
SIP/206-056c m...@agents:1Up  AppQueue((Outgoing
Line))
SIP/234-056b 1...@agents:1   Up  AppQueue((Outgoing
Line))
DAHDI/18-1   t...@zon:7  Up
Queue(Timeout-ZON,t,,,60)
DAHDI/4-1m...@fnacsaclojas:6  Up
Queue(FnacSACLojas,t,,,180)
SIP/208-0569 m...@agents:1Up  AppQueue((Outgoing
Line))
DAHDI/13-1   m...@fnacsaclojas:6  Up
Queue(FnacSACLojas,t,,,180)
DAHDI/30-1   1...@zon:38 Up
Queue(ZON,t,,,60)
SIP/227-0561 t...@agents:1   Up  AppQueue((Outgoing
Line))
DAHDI/24-1   t...@hf:9   Up
Queue(Timeout-HF,t,,,60)
SIP/233-0558 t...@agents:1   Up  AppQueue((Outgoing
Line))
SIP/216-0553 t...@agents:1   Up  AppQueue((Outgoing
Line))
DAHDI/20-1   t...@zon:7  Up
Queue(Timeout-ZON,t,,,60)
DAHDI/8-1t...@zon:7  Up
Queue(Timeout-ZON,t,,,60)
SIP/236-0545 t...@agents:1   Up  AppQueue((Outgoing
Line))
SIP/235-0541 t...@agents:1   Up  AppQueue((Outgoing
Line))
DAHDI/12-1   t...@zon:7  Up
Queue(Timeout-ZON,t,,,60)
DAHDI/6-1t...@zon:7  Up
Queue(Timeout-ZON,t,,,60)
SIP/219-0449 m...@agents:1Up  AppQueue((Outgoing
Line))
DAHDI/29-1   m...@fnacsaclojas:6  Up
Queue(FnacSACLojas,t,,,180)
24 active channels
13 active calls
3863 calls processed

The ael that is processed when a queue dials 611 looks like:

_XXX = {   // internal dial to extensions from queue.

Set(GROUP()=${EXTEN});  // increment group
count
Set(CDR(accountcode)=ext${ext});// for Phoenix
Set(OUTBOUND_GROUP=${EXTEN});   // same for channel
that will be created by Dial()
NoOp(GROUP_COUNT of ${EXTEN}: ${GROUP_COUNT(${EXTEN})});
if (${GROUP_COUNT(${EXTEN})} = 1)   // if not already in
call
{
Set(DIALSTART=${EPOCH});

Queue_log(${UNIQUEID},${PARTNER},${EXTEN},DIAL,${CALLERID(name)});

NoOp(PCmedicInfo: Followme seria:
followme/${PARTNER} - CallerID: ${CALLERID(number)} - UnID: ${UNIQUEID} -
Nao ha partner?... );
Set(NewCallMsg=followme/${PARTNER});
if (${NewCallMsg} = )
{
Set(NewCallMsg=followme/no-recording);
}
if (${NewCallMsg} = followme/)
{
Set(NewCallMsg=followme/no-recording); //
Geada - o IF anterior deveria verificar o PARNER?
NoOp(PCmedicInfo: Corrected followme: -
partner: ${PARTNER} - ${CALLERID(number)});
}
Dial(SIP/${EXTEN},7,rktgA(${NewCallMsg}));
if(${DIALSTART} != )
{
Set(DIALEDTIME=$[${EPOCH} - ${DIALSTART}]);
}
else
{
Set(DIALEDTIME=0);
}

ChanIsAvail(SIP/${EXTEN});  //
NoOp(DIALSTATUS EXT ${EXTEN}:); //
NoOp(DIALSTATUS=${DIALSTATUS});
//  Necessary for Phoenix
NoOp(DIALEDTIME=${DIALEDTIME}); //
NoOp(HANGUPCAUSE=${HANGUPCAUSE});   //
NoOp(AVAILSTATUS=${AVAILSTATUS});   //

if (${DIALSTATUS} 

Re: [asterisk-users] member (In use)

[Tiago Geada]

Because we already have a reduntant way to tell if the member is in a call,
we turned on ringinuse. It seems to work.

The member is still show as (In use).


Would anybody help?

Thanks.

2009/12/15 Tiago Geada tiago.ge...@gmail.com

 Hello list.

 We just upgraded to 1.6.1.11.

 We are using real time information stored on mysql databases. That is all
 running fine.

 Now, since we upgraded, some member don't get calls from queues.
 In CLI: queue show shows something like:
 611 (Local/6...@agents) with penalty 20 (realtime) (*In use*) has taken no
 calls yet


 We use the extension 611 in different computers, in the internal network
 with no nat, in the external network with nat...
 We deleted the member 611 from mysql, and added it again, changed passwd
 etc...
 We restarted asterisk several times..

 The member shows always (In use) !!

 Just to show that there is no channel associated with the member
 core show channels shows:
 Connected to Asterisk 1.6.1.11 currently running on perfpbxr (pid = 12955)
 Channel  Location State
 Application(Data)
 DAHDI/9-1m...@fnacsaclojas:2  Up
 Playback(audio/FnacSAC/qualida
 DAHDI/31-1   s...@zon:7  Up
 BackGround(audio/ZON/prima1)
 SIP/209-0570 m...@agents:1Up  AppQueue((Outgoing
 Line))
 SIP/604-056e t...@agents:1   Up  AppQueue((Outgoing
 Line))
 DAHDI/5-1m...@fnacsacbilhetei Up
 Queue(FnacSACBilheteira,t,,,18
 SIP/206-056c m...@agents:1Up  AppQueue((Outgoing
 Line))
 SIP/234-056b 1...@agents:1   Up  AppQueue((Outgoing
 Line))
 DAHDI/18-1   t...@zon:7  Up
 Queue(Timeout-ZON,t,,,60)
 DAHDI/4-1m...@fnacsaclojas:6  Up
 Queue(FnacSACLojas,t,,,180)
 SIP/208-0569 m...@agents:1Up  AppQueue((Outgoing
 Line))
 DAHDI/13-1   m...@fnacsaclojas:6  Up
 Queue(FnacSACLojas,t,,,180)
 DAHDI/30-1   1...@zon:38 Up
 Queue(ZON,t,,,60)
 SIP/227-0561 t...@agents:1   Up  AppQueue((Outgoing
 Line))
 DAHDI/24-1   t...@hf:9   Up
 Queue(Timeout-HF,t,,,60)
 SIP/233-0558 t...@agents:1   Up  AppQueue((Outgoing
 Line))
 SIP/216-0553 t...@agents:1   Up  AppQueue((Outgoing
 Line))
 DAHDI/20-1   t...@zon:7  Up
 Queue(Timeout-ZON,t,,,60)
 DAHDI/8-1t...@zon:7  Up
 Queue(Timeout-ZON,t,,,60)
 SIP/236-0545 t...@agents:1   Up  AppQueue((Outgoing
 Line))
 SIP/235-0541 t...@agents:1   Up  AppQueue((Outgoing
 Line))
 DAHDI/12-1   t...@zon:7  Up
 Queue(Timeout-ZON,t,,,60)
 DAHDI/6-1t...@zon:7  Up
 Queue(Timeout-ZON,t,,,60)
 SIP/219-0449 m...@agents:1Up  AppQueue((Outgoing
 Line))
 DAHDI/29-1   m...@fnacsaclojas:6  Up
 Queue(FnacSACLojas,t,,,180)
 24 active channels
 13 active calls
 3863 calls processed

 The ael that is processed when a queue dials 611 looks like:

 _XXX = {   // internal dial to extensions from queue.

 Set(GROUP()=${EXTEN});  // increment group
 count
 Set(CDR(accountcode)=ext${ext});// for Phoenix
 Set(OUTBOUND_GROUP=${EXTEN});   // same for channel
 that will be created by Dial()
 NoOp(GROUP_COUNT of ${EXTEN}: ${GROUP_COUNT(${EXTEN})});
 if (${GROUP_COUNT(${EXTEN})} = 1)   // if not already
 in call
 {
 Set(DIALSTART=${EPOCH});

 Queue_log(${UNIQUEID},${PARTNER},${EXTEN},DIAL,${CALLERID(name)});

 NoOp(PCmedicInfo: Followme seria:
 followme/${PARTNER} - CallerID: ${CALLERID(number)} - UnID: ${UNIQUEID} -
 Nao ha partner?... );
 Set(NewCallMsg=followme/${PARTNER});
 if (${NewCallMsg} = )
 {
 Set(NewCallMsg=followme/no-recording);
 }
 if (${NewCallMsg} = followme/)
 {
 Set(NewCallMsg=followme/no-recording); //
 Geada - o IF anterior deveria verificar o PARNER?
 NoOp(PCmedicInfo: Corrected followme: -
 partner: ${PARTNER} - ${CALLERID(number)});
 }
 Dial(SIP/${EXTEN},7,rktgA(${NewCallMsg}));
 if(${DIALSTART} != )
 {
 Set(DIALEDTIME=$[${EPOCH} - ${DIALSTART}]);
 }
 else
 {
 Set(DIALEDTIME=0);
 }

 ChanIsAvail(SIP/${EXTEN});  //
 NoOp(DIALSTATUS EXT ${EXTEN

Re: [asterisk-users] member (In use)

[Tiago Geada]

Because we already have a reduntant way to tell if the member is in a call,
we turned on ringinuse. It seems to work.

The member is still show as (In use).


Would anybody help?

Thanks.

2009/12/15 Tiago Geada tiago.ge...@gmail.com

 Hello list.

 We just upgraded to 1.6.1.11.

 We are using real time information stored on mysql databases. That is all
 running fine.

 Now, since we upgraded, some member don't get calls from queues.
 In CLI: queue show shows something like:
 611 (Local/6...@agents) with penalty 20 (realtime) (*In use*) has taken no
 calls yet


 We use the extension 611 in different computers, in the internal network
 with no nat, in the external network with nat...
 We deleted the member 611 from mysql, and added it again, changed passwd
 etc...
 We restarted asterisk several times..

 The member shows always (In use) !!

 Just to show that there is no channel associated with the member
 core show channels shows:
 Connected to Asterisk 1.6.1.11 currently running on perfpbxr (pid = 12955)
 Channel  Location State
 Application(Data)
 DAHDI/9-1m...@fnacsaclojas:2  Up
 Playback(audio/FnacSAC/qualida
 DAHDI/31-1   s...@zon:7  Up
 BackGround(audio/ZON/prima1)
 SIP/209-0570 m...@agents:1Up  AppQueue((Outgoing
 Line))
 SIP/604-056e t...@agents:1   Up  AppQueue((Outgoing
 Line))
 DAHDI/5-1m...@fnacsacbilhetei Up
 Queue(FnacSACBilheteira,t,,,18
 SIP/206-056c m...@agents:1Up  AppQueue((Outgoing
 Line))
 SIP/234-056b 1...@agents:1   Up  AppQueue((Outgoing
 Line))
 DAHDI/18-1   t...@zon:7  Up
 Queue(Timeout-ZON,t,,,60)
 DAHDI/4-1m...@fnacsaclojas:6  Up
 Queue(FnacSACLojas,t,,,180)
 SIP/208-0569 m...@agents:1Up  AppQueue((Outgoing
 Line))
 DAHDI/13-1   m...@fnacsaclojas:6  Up
 Queue(FnacSACLojas,t,,,180)
 DAHDI/30-1   1...@zon:38 Up
 Queue(ZON,t,,,60)
 SIP/227-0561 t...@agents:1   Up  AppQueue((Outgoing
 Line))
 DAHDI/24-1   t...@hf:9   Up
 Queue(Timeout-HF,t,,,60)
 SIP/233-0558 t...@agents:1   Up  AppQueue((Outgoing
 Line))
 SIP/216-0553 t...@agents:1   Up  AppQueue((Outgoing
 Line))
 DAHDI/20-1   t...@zon:7  Up
 Queue(Timeout-ZON,t,,,60)
 DAHDI/8-1t...@zon:7  Up
 Queue(Timeout-ZON,t,,,60)
 SIP/236-0545 t...@agents:1   Up  AppQueue((Outgoing
 Line))
 SIP/235-0541 t...@agents:1   Up  AppQueue((Outgoing
 Line))
 DAHDI/12-1   t...@zon:7  Up
 Queue(Timeout-ZON,t,,,60)
 DAHDI/6-1t...@zon:7  Up
 Queue(Timeout-ZON,t,,,60)
 SIP/219-0449 m...@agents:1Up  AppQueue((Outgoing
 Line))
 DAHDI/29-1   m...@fnacsaclojas:6  Up
 Queue(FnacSACLojas,t,,,180)
 24 active channels
 13 active calls
 3863 calls processed

 The ael that is processed when a queue dials 611 looks like:

 _XXX = {   // internal dial to extensions from queue.

 Set(GROUP()=${EXTEN});  // increment group
 count
 Set(CDR(accountcode)=ext${ext});// for Phoenix
 Set(OUTBOUND_GROUP=${EXTEN});   // same for channel
 that will be created by Dial()
 NoOp(GROUP_COUNT of ${EXTEN}: ${GROUP_COUNT(${EXTEN})});
 if (${GROUP_COUNT(${EXTEN})} = 1)   // if not already
 in call
 {
 Set(DIALSTART=${EPOCH});

 Queue_log(${UNIQUEID},${PARTNER},${EXTEN},DIAL,${CALLERID(name)});

 NoOp(PCmedicInfo: Followme seria:
 followme/${PARTNER} - CallerID: ${CALLERID(number)} - UnID: ${UNIQUEID} -
 Nao ha partner?... );
 Set(NewCallMsg=followme/${PARTNER});
 if (${NewCallMsg} = )
 {
 Set(NewCallMsg=followme/no-recording);
 }
 if (${NewCallMsg} = followme/)
 {
 Set(NewCallMsg=followme/no-recording); //
 Geada - o IF anterior deveria verificar o PARNER?
 NoOp(PCmedicInfo: Corrected followme: -
 partner: ${PARTNER} - ${CALLERID(number)});
 }
 Dial(SIP/${EXTEN},7,rktgA(${NewCallMsg}));
 if(${DIALSTART} != )
 {
 Set(DIALEDTIME=$[${EPOCH} - ${DIALSTART}]);
 }
 else
 {
 Set(DIALEDTIME=0);
 }

 ChanIsAvail(SIP/${EXTEN});  //
 NoOp(DIALSTATUS EXT ${EXTEN

[asterisk-users] monitor-type=MixMonitor

[Tiago Geada]

Hi!

Since we upgraded to 1.6.1.11, asterisk is only outputing monitored files
-in and -out.
It is not mixing them in the end.

queues.conf has monitor-type=MixMonitor...

Would somebody help me debug why it doesn't mix the sounds??

Thanks
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