Re: [asterisk-users] blacklist/V* - using wildcard

[Doug Lytle]

Joseph wrote:
So if the: "Set(goaway=${CALLERID(number):0:2})" limits the caller ID 
string to 2-characters, isn't it?


Actually, it hasn't doing anything to the caller ID, it's assigning the 
first 2 characters of the CID to the variable.


If I set it to 10-characters: ...${CALLERID(number):0:10} can I test 
it for the first two characters only "V4" or it has to be string 
10-characters long? 


No, that's what we're doing above.

Looking only at the first two characters of the CID and assigning it to 
${goaway} and then doing the testing on ${goaway}, leaving the CID intact.



Doug


--
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] sip registration

[Steve Edwards]

A better subject will yield better replies.

On Sat, 6 Apr 2013, Thomas Perron wrote:


Shouldnt I be able to at least ping the SIP provider IP?


Not if they don't allow it. They don't.

sip3.voipvoip.com registers fine for me with your credentials.

Did you put the registration statement in the [general] section?

I use the 'append' format so I can group all the cruft for a provider 
together. Like:


; voipvoip.com
[general](+)
register= 
nn:xxx...@sip3.voipvoip.com/nn
[outgoing]
secret  = xx
username= nn
...


I have not configured anything other then entries in the sip.conf


I used your credentials and successfully placed a call to all of my 
Caribbean premium numbers*.


Please change your password. Maybe your issue lies elsewhere. Does 
enabling SIP debugging on the console yield any clues?


*) just kidding.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] blacklist/V* - using wildcard

[Joseph]

On 04/06/13 14:38, Doug Lytle wrote:

Joseph wrote:

exten => s,4,GotoIf($["${CALLERIDNUM}" = "V4*"]?blacklisted,s,1


I don't believe so, you'd have to assign a variable to the first 2
characters and test that variable for V4.  i.e.

exten => s,1,Set(goaway=${CALLERID(number):0:2})
exten => s,n,GotoIf($["${goaway}" = "V4" ]?blacklisted,s,1)
exten => s,n,continue on with your normal processing

[blacklisted]

exten => s,1,Set(CDR(userfield)=Blacklisted)
exten => s,n,Zapateller(answer)
exten => s,n,Hangup(2)


Doug


Thank you! Yes, this is working :-/ 


So if the: "Set(goaway=${CALLERID(number):0:2})" limits the caller ID string to 
2-characters, isn't it?
If I set it to 10-characters: ...${CALLERID(number):0:10} can I test it for the first two 
characters only "V4" or it has to be string 10-characters long?

--
Joseph

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk SIP deadlocks - update_provisional_keepalive

[Duane Larson]

Looks like version 11.3 did not fix my issue.

http://pastebin.com/gd291Bqz


On Thu, Apr 4, 2013 at 1:23 PM, Duane Larson  wrote:

> Thanks Jim.  Searched through the change log for "deadlock" but nothing
> really stuck out.  I'll upgrade to 11.3 and see if that makes a difference.
>
>
> On Thu, Apr 4, 2013 at 10:59 AM, Jim Lucas  wrote:
>
>> On 04/03/2013 08:15 PM, Duane Larson wrote:
>>
>>> So it just happened again on both machines at the same time and I was
>>> running debug on both servers.  I am running OpenSIPS and load balancing
>>> between both servers so I am guessing when the invite was sent to the
>>> first
>>> server it was frozen for some reason and then OpenSIPS sent the invite to
>>> the second server and that server was also frozen/deadlocked because of
>>> the
>>> SIP message.  I noticed on both servers the last log that was posted with
>>> Asterisk deadlocked was the following
>>>
>>>
>>> Asterisk version 11.0.1
>>> [Apr  3 21:39:42] DEBUG[12984] res_timing_timerfd.c: Expected to
>>> acknowledge 1 ticks but got 11805 instead
>>>
>>> Asterisk version 11.2.1
>>> [Apr  3 21:39:50] DEBUG[1854] res_timing_timerfd.c: Expected to
>>> acknowledge
>>> 1 ticks but got 12423 instead
>>>
>>>
>>> In my last email I posted the debug from the Asterisk server with 11.0.1
>>> version of code.  Here is a post of the debug for the Asterisk server
>>> with
>>> version 11.2.1
>>>
>>> http://pastebin.com/mbjSSAWM
>>>
>>>
>>> This has to be a bug right?  I am thinking of opening an issue on the
>>> Asterisk JIRA system
>>>
>>>
>> A number of deadlocks were fixed in the current release of 11.3.  Please
>> read the change log to see if any fit your issue.
>>
>> http://downloads.asterisk.org/**pub/telephony/asterisk/**
>> ChangeLog-11-current
>>
>>
>>
>>>
>>> On Wed, Apr 3, 2013 at 4:45 PM, Duane Larson 
>>> wrote:
>>>
>>>  It just happened again on the 11.0.1 box and I was able to grab a debug.
   I am hoping someone can tell me if this is a bug or something wrong
 with
 my config.

 gdb asterisk-bin/sbin/asterisk 29048

 Go here for the debug output
 http://pastebin.com/DGXx0BSk


 On Tue, Apr 2, 2013 at 7:42 PM, Duane Larson >>> >wrote:

  I am currently running two different versions of Asterisk
>
> 11.0.1
> 11.2.1
>
> I have noticed the bug occur on both servers.
>
> The issue is that when I try to dial a phone number sometimes the call
> will never go out.  I will check the Asterisk server with NGREP and see
> that the SIP messages are making it to Asterisk but Asterisk isn't
> responding.
>
> I do the following command "netstat -nap |grep 5060" and see that
> Asterisk has a lot under the "Recv-Q" column.
>
> It usually takes about 10 minutes before Asterisk becomes responsive
> again or else before 10 minutes is up I could restart Asterisk and
> everything will be back to normal.
>
> I see in the message logs the following errors
>
> On the 11.0.1 Asterisk server
> WARNING[23723][C-0010] chan_sip.c: Unable to cancel schedule ID
> 11473.  This is probably a bug (chan_sip.c:
> update_provisional_keepalive,
> line 4406).
>
> On the 11.2.1 Asterisk server
> WARNING[3493][C-001f] chan_sip.c: Unable to cancel schedule ID
> 30810.
>   This is probably a bug (chan_sip.c: update_provisional_keepalive,
> line
> 4683).
>
>
> When I look in chan_sip.c on both servers I see that they are the same
> line of code
>
> AST_SCHED_DEL_UNREF(sched, pvt->provisional_keepalive_**sched_id,
> dialog_unref(pvt, "when you delete the provisional_keepalive_sched_**id,
> you
> should dec the refcount for the stored dialog ptr"));
>
>
>
> What could be causing this because it seems to happen at least once a
> day.
>
>


 --
 --
 *--*--*--*--*--*
 Duane
 *--*--*--*--*--*
 --


>>>
>>>
>>>
>>>
>>> --
>>> __**__**
>>> _
>>> -- 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
>>>
>>>
>>
>> --
>> Jim Lucas
>>
>> http://www.cmsws.com/
>> http://www.cmsws.com/examples/
>>
>
>
>
> --
> --
> *--*--*--*--*--*
> Duane
> *--*--*--*--*--*
> --
>



-- 
--
*--*--*--*--*--*
Duane
*--*--*--*--*--*
--
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introd

[asterisk-users] sip registration

[Thomas Perron]

I have a very lite layout and attempting to get the SIP configuration set
up initially before proceeding into other areas.

VMware is running my Asterisk 11 on Ubuntu 12.

Shouldnt I be able to at least ping the SIP provider IP?
I run command "sip show registry" and do not see it set up.
I run sip show peers and I do see an entry.

I have not configured anything other then entries in the sip.conf

results are:

Name/username HostDyn
Forcerport ACL Port Status  Description
outgoing/5552530146 (your
69.90.209.57   5060 OK (85
ms)
1 sip peers [Monitored: 1 online, 0 offline Unmonitored: 0 online, 0
offline]
Asterisk*CLI> sip show registry
Hostdnsmgr Username   Refresh
StateReg.Time
0 SIP registrations.
Asterisk*CLI>


my config is this:

[outgoing]
username=5552530146 (your VoIP VoIP account assigned while signing up)
type=peer
qualify=yes
secret=iblockedthis (your VoIP VoIP password)
nat=auto
insecure=invite,port
host=sip3.voipvoip.com
fromuser=5552530146 (your VoIP VoIP account assigned while signing up)
fromdomain=sip3.voipvoip.com
dtmfmode=rfc2833
disallow=all
allow=g729
allow=ilbc
allow=ulaw
allow=alaw
;
;
;
;
;
;register => 5552530146:7036361399@69.90.209.57/5552530146
register=>5552530146:boston!@#1...@sip3.voipvoip.com/5552530146
;


Please send input or guidance...

Thanks
Thomas
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] blacklist/V* - using wildcard

[Doug Lytle]

Joseph wrote:
exten => s,4,GotoIf($["${CALLERIDNUM}" = "V4*"]?blacklisted,s,1 


I don't believe so, you'd have to assign a variable to the first 2 
characters and test that variable for V4.  i.e.


exten => s,1,Set(goaway=${CALLERID(number):0:2})
exten => s,n,GotoIf($["${goaway}" = "V4" ]?blacklisted,s,1)
exten => s,n,continue on with your normal processing

[blacklisted]

exten => s,1,Set(CDR(userfield)=Blacklisted)
exten => s,n,Zapateller(answer)
exten => s,n,Hangup(2)


Doug


--
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] blacklist/V* - using wildcard

[Joseph]

On 04/06/13 11:59, Joseph wrote:

Does wildcard "*" character works in asterisk "blacklist"?

I have a telemarketing caller who is rotating caller id but they usually start with 
"V4-something" eg. V40611320600265
I've tried to block using whildcard "*" :
/blacklist/V4*   : advertising

but it doesn't work.


If I use in dialplan:
exten => s,4,GotoIf($["${CALLERIDNUM}" = "V4*"]?blacklisted,s,1 


will it work?

--
Joseph

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] blacklist/V* - using wildcard

[Joseph]

Does wildcard "*" character works in asterisk "blacklist"?

I have a telemarketing caller who is rotating caller id but they usually start with 
"V4-something" eg. V40611320600265
I've tried to block using whildcard "*" :
/blacklist/V4*   : advertising

but it doesn't work.

--
Joseph

--
_
-- 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