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  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 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  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] 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  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  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 
>
> 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
> ] *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] 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  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  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  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 :
>
> 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 :
>>
>> 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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>>
>>
>
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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  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  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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>
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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  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] 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  wrote:

>
>
>
> On Fri, Jun 27, 2014 at 1:30 PM, Tiago Geada 
> 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] 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  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  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  wrote:
>
>>
>>
>>
>> On Wed, Jun 11, 2014 at 9:10 AM, Tiago Geada 
>> 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:
>>
>> 
>>
>>
>>>
>>>>
>>>>
>>>>
>>>> ## 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
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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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  wrote:

>
>
>
> On Wed, Jun 11, 2014 at 9:10 AM, Tiago Geada 
> 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:
>
> 
>
>
>>
>>>
>>>
>>>
>>> ## 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
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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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  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   
>    Set channel variable  to 
>
>
>
>
>
> On 10 June 2014 16:39, Tiago Geada  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();
>
> }
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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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  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
> --
> This email is made from 100% recycled electrons
>
>
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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  wrote:

> On Mon, Jan 27, 2014 at 1:02 PM, Ron Wheeler
>  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
>
> --
> 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] 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  wrote:

> Gareth,
>
> I had to disable the tcpdump process, has we were having sound quality
> issues.
>
> :-(
>
>
> On 16 January 2014 15:35, Gareth Blades 
> wrote:
>
>>  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
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
>
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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 wrote:

>  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]

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


On 16 January 2014 15:29, Kevin Larsen wrote:

> asterisk-users-boun...@lists.digium.com wrote on 01/16/2014 08:55:31 AM:
>
> > From: Gareth Blades 
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > ,
> > 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]

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 wrote:

>  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 $callID' -w - |
> ngrep -I - -W byline -t
>
>
>
> On 16/01/14 14:57, Tiago Geada wrote:
>
>  Second thought, that would only allow me to know if there is a problem
> on asterisk or softphone.
>
>  Because the old callerid(name) that was presented on the softphone,
> belonged to a call made to a different peer, I doubt that it would be a
> softphone problem.
>
>  Our softphones are fixed with the same peer/extension .. if the wrong
> callerid were originally called to the same peer.. I guess that would be
> worth it..
>
>  even so, I will try and measure the impact on performance, however if
> asterisk did send the wrong string, how could I debug that??
>
>
> On 16 January 2014 14:27, Tiago Geada  wrote:
>
>>  You're right, seems like a nice way to debug. Regarding that, how would
>> the impact be affected running it on asterisk box? I guess only port 5060
>> is not too bad
>>
>>
>>  On 16 January 2014 14:09, Gareth Blades <
>> mailinglist+aster...@dns99.co.uk> wrote:
>>
>>>On 16/01/14 10:47, Tiago Geada wrote:
>>>
>>>  Hi folks,
>>>
>>>  We've been having a weird issue... It is happening more often in the
>>> last few months...
>>>
>>>  Most inbound calls, we have in our dialplan before Queue():
>>>
>>>
>>> Set(CALLERID(name)=${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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>>>
>>
>>
>
>
>
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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 wrote:

>  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 $TRACEFILE -C 10 -W
> 500 &
>
>
>
> On 16/01/14 14:27, Tiago Geada wrote:
>
>  You're right, seems like a nice way to debug. Regarding that, how would
> the impact be affected running it on asterisk box? I guess only port 5060
> is not too bad
>
>
> On 16 January 2014 14:09, Gareth Blades 
> wrote:
>
>>   On 16/01/14 10:47, Tiago Geada wrote:
>>
>>  Hi folks,
>>
>>  We've been having a weird issue... It is happening more often in the
>> last few months...
>>
>>  Most inbound calls, we have in our dialplan before Queue():
>>
>>  Set(CALLERID(name)=${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.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
>
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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  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 
> 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.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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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 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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[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] 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  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/something&Local/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] Caller ID not real nor showing in call logs.

[Tiago Geada]

logs ?

full log containing the call?


On 8 January 2014 14:56, Eherr  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] 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  wrote:

> On Wed, Dec 11, 2013 at 10:20 PM, Tzafrir Cohen
>  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.
>
>
> --
> Dotan Cohen
>
> 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  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"  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  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  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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>
>
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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  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 1000; 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  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
> 1000; 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 1000
>>> ;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 \'\' - Wrong password
>> Unauthenticated call attempt .*\@\:
>>
>>
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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  wrote:

>
>
>
>   On Wednesday, September 11, 2013 1:54 PM, longst 
> 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  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] 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  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] 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  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  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  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] 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  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
>>
>> --
>> __**__**_
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>> http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>> 
>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>
>>
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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  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] 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  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] 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  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] 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  wrote:

>
>
> On 25/11/2012, at 1:23 PM, Tiago Geada  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=1&t=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  wrote:
>
>>
>> On 24/11/2012, at 2:19 AM, Tiago Geada  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  wrote:

> Its depends on dialplan and the way you treat the call.
>
>
> On Fri, Oct 26, 2012 at 7:54 PM, Mitch Claborn wrote:
>
>> 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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>
>
>
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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  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  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  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"  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 
>Systems Engineer, OfficeWyze, Inc.
> Ph: +1 415 439 4988   Fax: +1 415
> 283 3370 
>http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20 <
> http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20>
>
>
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>
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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  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: 
>
> Indirect** **Bridge: 
>
>  --   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

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  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 
> Monitoror
>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  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
> >  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  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
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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  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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Re: [asterisk-users] estimated queue hold time

[Tiago Geada]

Hi again,


Perhaps I could execute a macro while joining the queue and evaluate
the QUEUEHOLDTIME variable there? If so can I in the macro remove a member
from the queue it just joined ???


thanks in advance


On 5 May 2011 19:41, Tiago Geada  wrote:

> 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] receive faxes

[Tiago Geada]

spandsp works fine on our PRI line

context FAX
{
s => {
NoOp(INFO: Getting fax in );
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/bin/sendEmail -f aster...@xxx.pt -t
f...@xxx.pt-u "Novo fax recebido" -a
/var/log/asterisk/fax/fax.${UNIQUEID}.tif -m
"Conteudo:" -s mail.xxx.pt:25 -xu asterisk@$
System(/usr/local/bin/faxmail "${recordPath}"
"${recordFile}");
};
//  sendfax => {
//  SendFax(/var/log/asterisk/fax/fax.1247153669.3.tif);
//  };
}


On 5 May 2011 19:09, David Backeberg  wrote:

> On Thu, May 5, 2011 at 1:43 PM, vip killa  wrote:
> > The majority of open source projects out are NOT run by commercial
> > institutions...
>
> Postfix kicks butt. But only because IBM paid for development, for a
> long number of years, and because they hired somebody who had a really
> good idea how to improve Sendmail.
>
> Asterisk kicks butt. It just does, even if you can't get your silly
> faxing edge case to work, because Digium paid for development, for a
> long number of years.
>
> There are also people who come along, and because the source is
> available, fix a particular bug that annoyed them, or added a feature
> they personally wanted.
>
> T.38 has a boatload of problems, and most of those problems are
> because people who aren't employed by Digium did not read the specs,
> or they did read the specs, but felt like they had to violate the
> specs to get their code to work with a different broken T.38 stack.
>
> I've personally fixed problems with asterisk, and I found my code
> contribution accepted. Perhaps you've submitted patches the bug
> tracker? We eagerly anticipate your voluntary code contributions.
>
> Maybe statistically, there are more open source projects out there
> that have non-paid lead developers, and they do their work on their
> own dime, and on their own time.
>
> But unless you're lucky enough to live in a place where you can hunt
> and gather for everything you need, you need money to live in a
> society, and have the resources to be able to sit at a keyboard long
> enough to churn out code.
>
> I could only afford to make my 'code contribution' because I have a
> day job where I needed the fix.
>
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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] Best modem for chan_datacard

[Tiago Geada]

I used succesfully huawei E1550

On 24 April 2011 16:45, Dovid Bender  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] 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  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/
>
> --
> Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
> 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
>
>
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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  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:-*
>
>  $priline = system('/usr/sbin/asterisk -rnx "pri show spans"',$pri);
> $asterisk = system("/etc/init.d/asterisk status", $asterisks);
> $mysql = system("/etc/init.d/mysql status",$mysqls);
> echo "priline=>".$priline;
> echo "";
> echo "pri=>".$pri;
> echo "";
> echo "asterisk=>".$asterisk;
> echo "";
> echo "asterisks=>".$asterisks;
> echo "";
> echo "mysql=>".$mysql;
> echo "";
> echo "mysqls=>".$mysqls;
> echo "";
> ?>
>
> *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 wrote:
>
>> Hi:
>>
>> http://php.net/manual/en/function.system.php
>>
>> Then, the commands you shoul run:
>>
>> /usr/sbin/asterisk -rnx"pri 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 wrote:
>>
>>> 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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>>>
>>
>>
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>>
>
>
>
> --
>
>
>
> -
>
> Thanks and regards
>
>  Virendra Bhati
> +91-9172341457
>
>
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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  wrote:

>
>
> On Sat, Apr 23, 2011 at 11:20 AM, Tiago Geada 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
>>
>>
> I'm a little confused by "It Seems I cannot re-use this var in extension
> _888XX in context agents"Of 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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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  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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[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] 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  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=, relay=asterisk@localhost
> Apr 11 12:57:57 shirley sendmail[29698]: p3BJvvtp029698: to="Satish Patel"
> , 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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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  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] 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  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  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  wrote:
>>
>>> On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini
>>>  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:
>>>   http://www.asterisk.org/hello
>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>
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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  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  wrote:
>
>> On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini
>>  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:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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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  wrote:

> On Fri, Mar 18, 2011 at 3:17 PM, Adrian Serafini
>  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:
>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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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  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
&g

[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 s

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.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
> --
> _
> -- 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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>
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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  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
>
> --
> Barry
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Queue Agent Getting Additional Calls When on the Phone

[Tiago Geada]

Hi,

We have realtime queues, and I can't figure how the device state matters,
because when a user takes a call, is stat is "Not in use".

We now use GROUP_COUNT() to check if the peer has a call or not...

On 15 October 2010 12:21, Leif Madsen  wrote:

> On 10-10-15 04:10 AM, Сикорский Сергей wrote:
> > 15.10.2010 9:40, Warren Selby пишет:
> >> I think this means you need to set a call-limit for each sip peer
> >
> > Is there any alternative for obsolete call-limit option in 1.6/1.8?
>
> The correct answer is to use ringinuse=no in queues.conf and
> callcounter=yes in
> sip.conf.
>
> Leif.
>
> --
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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  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  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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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  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 
>> 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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[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] 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  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 
> 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  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  > <mailto:paul.belan...@polybeacon.com>> wrote:
> >
> > On Mon, Jun 28, 2010 at 10:04 PM, Zhang Shukun  > <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  wrote:

> On Mon, Jun 28, 2010 at 10:04 PM, Zhang Shukun  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] 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  wrote:

> anyone know something about this?
>
>
> On Fri, May 14, 2010 at 10:56 AM, Jean Chassoul wrote:
>
>> 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] 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  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 :
> > 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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> >
> > 
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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  wrote:

> 2010/6/22 Philipp von Klitzing :
> > 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
> >
>
>
> --
> Thanks for your supporting,
> have a nice day.
> Sucan
>
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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 
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

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  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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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 

> 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] 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  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] Asterisk on Ubuntu

[Tiago Geada]

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

On 4 June 2010 13:40, Danny Dias  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] problem with inserting records into cdr

[Tiago Geada]

pasting the error would help

On 3 June 2010 20:56,  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] 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  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: "

sleep 0

echo "RCPT TO: "

echo "RCPT TO: "

sleep 1

echo "data"


echo "Subject: FAX $2"

echo "FROM: "

echo "TO: "

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)$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  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  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 

> 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
>  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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Re: [asterisk-users] Script to show asterisk stuff

[Tiago Geada]

2010/1/4 Will Payne 

>
> 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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[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] 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 

> 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] 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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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 

> 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(S