Re: [asterisk-users] how to flush user input before READ()

2016-01-18 Thread Ethy H. Brito 

Hi All.

No one ? Anyone?

Cheers

Ethy


On Fri, 15 Jan 2016 15:37:51 -0200
"Ethy H. Brito"  wrote:

> 
> Hi
> 
> how to flush user input before READ()?
> 
> I wrote a small script to ask for user password before granting access to
> outside, but some telefones memorize the full user input, including "#".
> 
> So, when the user press redial, for instance 5556789#123, asterisk accepts the
> number and the password "123" and gives access to the outside word to whomever
> redials that terminal.
> 
> Any hints?
> 
> Cheers
> 
> Ethy
> 
> -- 
> _
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-- 

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S.J.Campos - Brasil   / \ 
 
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Re: [asterisk-users] How to get PJSIP SIP messages in a log file and not in console ?

2016-01-18 Thread Olivier 
2016-01-18 13:53 GMT+01:00 Olivier :

> Hello,
>
> How should I configure Asterisk (13.7.0) to get persistently PJSIP SIP
> messages in a log file and not in console ?
>
> I would expect adding "debug=yes" in pjsip.conf to produce the same output
> as "pjsip set logger on".
> Am I understanding correctly ?
>

I've got:
> logger show channels
Channel Type StatusConfiguration
---  ---
/var/log/asterisk/messages  File Enabled- NOTICE WARNING
ERROR
Console  Enabled- NOTICE WARNING
ERROR
/var/log/asterisk/debug File Enabled- DEBUG NOTICE
WARNING ERROR


> pjsip show settings

Global Settings:

 ParameterName : ParameterValue
 =
 debug : yes




I expected to have PJSIP's SIP messages logged to /var/log/asterisk/debug
file and not find them in console.
I'm seeing the opposite (no SIP message in debug file and console cluttered
with SIP messages).
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Re: [asterisk-users] Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".

2016-01-18 Thread Jonathan H 
Gh! 15 minutes after reading your answer, I had it working perfectly!

Thank you!

Before I type it up, here's what works for me - can you see any
obvious flaws or hidden dangers here?

-
pjsip.conf
-

[acl]
type = acl
deny = 0.0.0.0/0.0.0.0
permit = 81.23.228.129,81.23.228.150,85.17.186.7

-
pjsip_wizard.conf
-

[sip2sipusername]
type = wizard
sends_auth = yes
sends_registrations = yes
remote_hosts = 81.23.228.129,85.17.186.7,81.23.228.150,sip2sip.info
outbound_auth/username = sip2sipusername
outbound_auth/password = sip2sippassword
endpoint/allow = alaw
endpoint/context = sip2sipusername
registration/contact_user = sip2sipusername
outbound_proxy = proxy.sipthor.net
endpoint/language=en_GB

-
extensions.conf
-

[sip2sipusername]
exten => sip2sipusername,1,NoOp()
same => n, playback(hello-world)

-

Again, thank you so much. I wish I'd discovered this mailing list
weeks ago - I'd assumed it was like those mailing lists that were "in
sync" with the forum.

Lesson learnt, and again, thank you.

On 18 January 2016 at 11:57, Joshua Colp  wrote:
> Jonathan H wrote:
>>
>> Would greatly appreciate any input into this currently-unanswered
>> question on the forum:
>>
>> http://forums.asterisk.org/viewtopic.php?f=1=96496
>>
>> I posted it on Jan 6th, have tried so many things, so much forum/list
>> searching and late nights since, but have had to admit defeat.
>>
>> Rather than duplicate it all here, I've posted my logs and conf files
>> on that thread, too.
>>
>> Problem is that while there are quite a few sip examples, I have
>> chosen to take the path of pjsip.
>>
>> Seems I can manage to attach Blink, Zoiper, Microsip and my ITSP with
>> multiple extensions without problem to my Asterisk, but sip2sip has
>> beaten me!
>>
>> It's presumably something ridiculously simple, but there comes a point
>> where you can't see the wood for the trees.
>>
>> If someone can help me resolve this, I'll post a complete guide on
>> Github Gist to help others in the future.
>
>
> It is likely that the IP address that traffic is coming from differs from
> the IP address resolved by res_pjsip_endpoint_identifier_ip. Currently that
> module is dumb and just does an A record lookup, it does not do any SRV or
> NAPTR lookup (which sip2sip likely uses). As a result when the INVITE comes
> in it does not identify it. You will need to determine the possible IP
> addresses and create your own identify section to match on them as the
> correct endpoint (I don't use wizards so don't know how to configure it with
> them).
>
> The current IP addresses possible being the following:
>
> proxy.sipthor.net.  60  IN  A   81.23.228.129
> proxy.sipthor.net.  60  IN  A   85.17.186.7
> proxy.sipthor.net.  60  IN  A   81.23.228.150
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.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] How to get PJSIP SIP messages in a log file and not in console ?

2016-01-18 Thread Olivier 
Changing my logger.conf file, I could get this:
I> logger show channels
Channel Type StatusConfiguration
---  ---
/var/log/asterisk/messages  File Enabled- NOTICE WARNING
ERROR
Console  Enabled- NOTICE WARNING
ERROR
/var/log/asterisk/pjsip File Enabled- VERBOSE

With this on, I've got PJSIP SIP messages in file /var/log/asterisk/pjsip
(great !) but I also have them showing at console (strange !).

Thoughts ?

2016-01-18 14:36 GMT+01:00 Olivier :

>
>
> 2016-01-18 13:53 GMT+01:00 Olivier :
>
>> Hello,
>>
>> How should I configure Asterisk (13.7.0) to get persistently PJSIP SIP
>> messages in a log file and not in console ?
>>
>> I would expect adding "debug=yes" in pjsip.conf to produce the same
>> output as "pjsip set logger on".
>> Am I understanding correctly ?
>>
>
> I've got:
> > logger show channels
> Channel Type StatusConfiguration
> ---  ---
> /var/log/asterisk/messages  File Enabled- NOTICE WARNING
> ERROR
> Console  Enabled- NOTICE WARNING
> ERROR
> /var/log/asterisk/debug File Enabled- DEBUG NOTICE
> WARNING ERROR
>
>
> > pjsip show settings
>
> Global Settings:
>
>  ParameterName : ParameterValue
>  =
>  debug : yes
>
>
>
>
> I expected to have PJSIP's SIP messages logged to /var/log/asterisk/debug
> file and not find them in console.
> I'm seeing the opposite (no SIP message in debug file and console
> cluttered with SIP messages).
>
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[asterisk-users] best practices - ari reconnect

2016-01-18 Thread Marek Červenka 

hi,

can you share your best practices for ARI reconnect when asterisk is 
restarted or when ari app is started before asterisk is fullybooted?


we are using node.js + ari-client so we are thinking about these options:
1) wait for AMI event FullyBooted
2) wait for AMI reconnect and then run ARI reconnect

thanks

--
---
Marek Cervenka
===


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Re: [asterisk-users] how to flush user input before READ()

2016-01-18 Thread Steve Edwards 

On Mon, 18 Jan 2016, Ethy H. Brito wrote:


how to flush user input before READ()?


How about a read() to a dummy variable with a 1 second timeout to consume
the octothorpe and password?

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

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Re: [asterisk-users] how to flush user input before READ()

2016-01-18 Thread Ethy H. Brito 
On Mon, 18 Jan 2016 09:38:52 -0800 (PST)
Steve Edwards  wrote:

> On Mon, 18 Jan 2016, Ethy H. Brito wrote:
> 
> >> how to flush user input before READ()?
> 
> How about a read() to a dummy variable with a 1 second timeout to consume
> the octothorpe and password?


It is an odd solution but i'll give it a try.

Thanx;

Ethy


> 
> -- 
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> 
> -- 
> _
> -- 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


-- 

Ethy H. Brito /"\
InterNexo Ltda.   \ /  CAMPANHA DA FITA ASCII - CONTRA MAIL HTML
+55 (12) 3797-6860 X   ASCII RIBBON CAMPAIGN - AGAINST HTML MAIL
S.J.Campos - Brasil   / \ 
 
PGP key: http://www.inexo.com.br/~ethy/0xC3F222A0.asc

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Re: [asterisk-users] PJSIP Returning 421 Extension Required

2016-01-18 Thread Matthew Jordan 
On Wed, Jan 13, 2016 at 12:58 PM, Trey Hilyard  wrote:

> I am turning up a PJSIP Endpoint and am having problems when they send an
> INVITE to my server. Asterisk is returning a 421 Extenstion Required. Since
> "extension" means different things in the SIP stack versus Asterisk, I
> don't know what it is complaining about.
>
> I have attached the trace below. Nothing else shows up with core verbose
> or core debug enabled, so I am assuming it has to be dying at the PJSIP
> module. The INVITE does come from an abnormal UDP Port, which is also shown
> in the Via header, but the fact that the PBX is responding makes me think
> that isn't the culprit.
>
> Any thoughts?
>
> SIP Logger:
> INVITE sip:+18165116504@12.4.240.200:5060;user=phone SIP/2.0
> v: SIP/2.0/UDP 10.77.27.103:20065
> ;branch=z9hG4bK0020C575A392E895C39051;oc-accept
> Max-Forwards: 70
> t: 
> f: ;tag=10847511385389740959
> i: 117620342110831512016142@10.77.27.103
> CSeq: 1 INVITE
> d: no-fork
> Privacy: none
> P-Asserted-Identity:
> 
> Require: 100rel
> Accept: application/sdp
> k: histinfo,resource-priority
> c: application/sdp
> m: 
> Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,UPDATE
> l:   228
>
> v=0
> o=PVG 1452710812870 1452710812870 IN IP4 10.77.160.55
> s=-
> c=IN IP4 10.77.160.55
> t=0 0
> m=audio 37700 RTP/AVP 0 101
> b=AS:80
> b=RR:0
> b=RS:0
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=maxptime:20
>
> <--- Transmitting SIP response (495 bytes) to UDP:10.77.27.103:20065 --->
> SIP/2.0 421 Extension Required
> Via: SIP/2.0/UDP 10.77.27.103:20065
> ;rport=20065;received=10.77.27.103;branch=z9hG4bK0020C575A392E895C39051;oc-accept
> Call-ID: 117620342110831512016142@10.77.27.103
> From:  ;user=phone>;tag=10847511385389740959
> To:  ;user=phone>;tag=z9hG4bK0020C575A392E895C39051
> CSeq: 1 INVITE
> Require: 100rel
> Supported: 100rel, timer, replaces, norefersub
> Server: Asterisk PBX 13.3.0-rc1
> Content-Length:  0
>
>
PJSIP is rejecting the inbound INVITE request as 100rel is required, but is
not in the Supported header of the inbound SIP INVITE request. I would
suspect that the UAC is doing things incorrectly by placing 100rel in the
Require but not in the list of option tags in the Supported header.

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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[asterisk-users] Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".

2016-01-18 Thread Jonathan H 
Would greatly appreciate any input into this currently-unanswered
question on the forum:

http://forums.asterisk.org/viewtopic.php?f=1=96496

I posted it on Jan 6th, have tried so many things, so much forum/list
searching and late nights since, but have had to admit defeat.

Rather than duplicate it all here, I've posted my logs and conf files
on that thread, too.

Problem is that while there are quite a few sip examples, I have
chosen to take the path of pjsip.

Seems I can manage to attach Blink, Zoiper, Microsip and my ITSP with
multiple extensions without problem to my Asterisk, but sip2sip has
beaten me!

It's presumably something ridiculously simple, but there comes a point
where you can't see the wood for the trees.

If someone can help me resolve this, I'll post a complete guide on
Github Gist to help others in the future.

Thanks.

-- 
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Re: [asterisk-users] Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".

2016-01-18 Thread Joshua Colp 

Jonathan H wrote:

Would greatly appreciate any input into this currently-unanswered
question on the forum:

http://forums.asterisk.org/viewtopic.php?f=1=96496

I posted it on Jan 6th, have tried so many things, so much forum/list
searching and late nights since, but have had to admit defeat.

Rather than duplicate it all here, I've posted my logs and conf files
on that thread, too.

Problem is that while there are quite a few sip examples, I have
chosen to take the path of pjsip.

Seems I can manage to attach Blink, Zoiper, Microsip and my ITSP with
multiple extensions without problem to my Asterisk, but sip2sip has
beaten me!

It's presumably something ridiculously simple, but there comes a point
where you can't see the wood for the trees.

If someone can help me resolve this, I'll post a complete guide on
Github Gist to help others in the future.


It is likely that the IP address that traffic is coming from differs 
from the IP address resolved by res_pjsip_endpoint_identifier_ip. 
Currently that module is dumb and just does an A record lookup, it does 
not do any SRV or NAPTR lookup (which sip2sip likely uses). As a result 
when the INVITE comes in it does not identify it. You will need to 
determine the possible IP addresses and create your own identify section 
to match on them as the correct endpoint (I don't use wizards so don't 
know how to configure it with them).


The current IP addresses possible being the following:

proxy.sipthor.net.  60  IN  A   81.23.228.129
proxy.sipthor.net.  60  IN  A   85.17.186.7
proxy.sipthor.net.  60  IN  A   81.23.228.150

Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org


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[asterisk-users] How to get PJSIP SIP messages in a log file and not in console ?

2016-01-18 Thread Olivier 
Hello,

How should I configure Asterisk (13.7.0) to get persistently PJSIP SIP
messages in a log file and not in console ?

I would expect adding "debug=yes" in pjsip.conf to produce the same output
as "pjsip set logger on".
Am I understanding correctly ?

Best regards
-- 
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[asterisk-users] Segmentation Fault Asterisk 13.7.0-rc2 (libmysqlclient?)

2016-01-18 Thread Matthew Murphy 
Hi everyone,

I am getting a segmentation fault (seems to occur randomly) using Asterisk 
13.7.0-rc2 with PJProject 2.4.5. It appears to be something that libmysqlclient 
is complaining about when doing a query in ps_endpoint_id_ips. We are using 
Asterisk Realtime. This also seems to occur in Asterisk 13.5.0.


Below is a backtrace that might help a little. I have looked through the change 
log for the 13.7.0 release and some of items addressed may fix my problem. 
Before diving in and attempting to upgrade to the final version of 13.7.0, I 
was hoping someone with knowledge would be able to look at this and let me know 
if this is something already seen or if this is entirely new.


Thanks for the help!


--Matt


---
BACKTRACE BELOW
---




[Thread debugging using libthread_db enabled]
Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1".
Core was generated by `asterisk -g'.
Program terminated with signal SIGSEGV, Segmentation fault.
#0  0x7f1e02e8a120 in list_add () from 
/usr/lib/x86_64-linux-gnu/libmysqlclient.so.18
#0  0x7f1e02e8a120 in list_add () from 
/usr/lib/x86_64-linux-gnu/libmysqlclient.so.18
No symbol table info available.
#1  0x7f1e0339d132 in my_SQLAllocStmt () from 
/usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so
No symbol table info available.
#2  0x7f1e38354af4 in ?? () from /usr/lib/x86_64-linux-gnu/libodbc.so.2
No symbol table info available.
#3  0x7f1e01e8f16f in custom_prepare (obj=0x1e588c8, data=0x7f1dc8a289b0) 
at res_config_odbc.c:107
res = 0
x = 1
count = 0
cps = 0x7f1dc8a289b0
field = 0x1e588a8
encodebuf = 
"\001\200\255\373\000\000\000\001P\205\242\310\035\177\000\000P\205\242\310\035\177\000\000P\205\242\310\035\177\000\000P\205\242\310\035\177\000\000\206\205\242\310\035\177\000\000\237\205\242\310\035\177\000\000P\205\242\310\035\177\000\000\237\205\242\310\035\177\000\000\342X\350\002\036\177\000\000\247p\350\001\000\000\000\000\351ʎ:\036\177\000\000\000\000\000\000\000\000\000\000\335Y\350\002\036\177\000\000`\000\000\000\004\000\000\000\a\000\000\000\000\000\000\000@\256\350\001\000\000\000\000\a",
 '\000' , 
"\340U\347\002\036\177\000\000\a\000\000\000\000\000\000\000;\000\000\000\000\000\000\000\002\000\000\000\035\177\000\001`\204\242\310\035\177\000\000\377\377\377\377\000\000\000\000\000"...
stmt = 0x0
__PRETTY_FUNCTION__ = "custom_prepare"
#4  0x7f1e385b9783 in ast_odbc_prepare_and_execute (obj=0x1e588c8, 
prepare_cb=0x7f1e01e8f11f , data=0x7f1dc8a289b0) at 
res_odbc.c:640
res = 0
i = 0
attempt = 0
nativeerror = 0
numfields = 0
diagbytes = 0
state = "\v:\351\001\036\177\000\000t\001"
diagnostic = 
"\340\030\006\244\035\177\000\000̜\242\310\035\177\000\000\200\020\024\002\000\000\000\000,Z_9\036\177\000\000)\266\227V\000\000\000\000\320\003\000\000\000\000\000\000\240\210\242\310\035\177\000\000\350\341\000D\035\177\000\000\300s\276\314\035\177\000\000\255I^\000\000\000\000\000\350\341\000D\035\177\000\000p>d\000\000\000\000\000̜\242\310\034\b\000\000K6d\000\000\000\000\000\320\003\000\000\000\000\000\000\001\000\000\000\000\000\000\000`\211\242\310\035\177\000\000Г\002\244\035\177\000\000\360\210\242\310\035\177\000\000\065\200^\000\000\000\000\000@F\351\001\036\177\000\000`\211\242\310t\001\000\000\v:\351\001\036\177\000\000\000\001\000\000\000\000\000\000\300\211\242\310\035\177\000\000ȑ\242\310"...
stmt = 0x7f1dc8a297b0
__PRETTY_FUNCTION__ = "ast_odbc_prepare_and_execute"
#5  0x7f1e01e90715 in realtime_multi_odbc (database=0x7f1dc8a298d0 
"asterisk", table=0x7f1dc8a297d0 "ps_endpoint_id_ips", fields=0x7f1da40185f0) 
at res_config_odbc.c:376
obj = 0x1e588c8
stmt = 0x25ed6b2
sql = "SELECT * FROM ps_endpoint_id_ips WHERE id LIKE ? ORDER BY 
id\000\177\000\000\377\377\377\377\377\377\377\377\000\000\000\000\000\000\000\000`\224\242\310\035\177\000\000m\225\242\310\035\177\000\000`\225\242\310\035\177\000\000`\226\242\310\035\177\000\000@\225\242\310\035\177\000\000\033\311d9\036\177\000\000P\225\242\310\035\177\000\000ȕ\242\310",
 '\000' , "\377\377\377\377", '\000' , " 
R\234\317\035\177\000\000\000\000\000\000\035\177", '\000' ...
coltitle = "\002", '\000' , 
"|\000\000\000\000\000\000\000ǫ\b\316", '\000' , "\002", 
'\000' , 
"\021\000\000\000\000\000\000\000\035\177\000\000\000\000\000\000\000\000\000\000
 
\000\000\000\000\062\062\062\000\000\000\000\035\177\000\000\000\000\000\000\035\177\000\000\000\000\000\000\035\177\000\000\000\000\000\000\036\177\000\000\377\377\377\377\035\177\000\000\000\000\000\000\036\177",
 '\000' , "\377\377\377\377\377\377\377\377%", '\000' 
, 
"\372I\\9\036\177\000\000\000\224\242\310\035\000\000\000\062"...
rowdata = 0x7f1da4001110
initfield = 0x7f1dc8a28960 "id"
op = 0x7f1e01e93b2d ""
field = 0x0