[asterisk-users] Qualify=yes and queues

[Антон Сацкий]

Hi list got  small question

for example i got 1  queue and 3  agents
but  there is no qualify in sip.conf for this users


IS there some influence  if I will use  qualify or not   on queue

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Antony
tel.   +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy

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Re: [asterisk-users] fail2ban Asterisk 13.13.1

[Антон Сацкий]

Think that U should ask in Fain2ban LIST

2017-03-01 20:29 GMT+02:00 Motty Cruz :

> Hello, fail2ban does not ban offending IP.
>
>
>
> NOTICE[29784] chan_sip.c: Registration from 
> '"user3"'
> failed for 'offending-IP:53417' - Wrong password
>
> NOTICE[29784] chan_sip.c: Registration from 
> '"user3"'
> failed for ‘offending-IP:53911' - Wrong password
>
>
>
> systemctl status fail2ban
>
> ● fail2ban.service - Fail2Ban Service
>
>Loaded: loaded (/usr/lib/systemd/system/fail2ban.service; enabled;
> vendor preset: disabled)
>
>Active: active (running) since Wed 2017-03-01 00:40:43 PST; 470min ago
>
>  Docs: man:fail2ban(1)
>
>
>
> jail.local
>
> [DEFAULT]
>
> # "bantime" is the number of seconds that a host is banned.
>
> bantime  = -1
>
>
>
> # A host is banned if it has generated "maxretry" during the last
> "findtime"
>
> # seconds.
>
> findtime  = 300
>
>
>
> # "maxretry" is the number of failures before a host get banned.
>
> maxretry = 3
>
>
>
> [asterisk-iptables]
>
> enable = true
>
> port = 5060,5061
>
> filter   = asterisk
>
> action   = iptables-allports[name=ASTERISK, protocol=all]
>
>   sendmail[name=ASTERISK, dest=mo...@email.com, sender=
> fail2...@asterisk-ip.com]
>
> #action   = %(banaction)s[name=%(__name__)s-tcp, port="%(port)s",
> protocol="tcp", chain="%(chain)s", actname=%(banaction)s-tcp]
>
>%(banaction)s[name=%(__name__)s-udp, port="%(port)s",
> protocol="udp", chain="%(chain)s", actname=%(banaction)s-udp]
>
>%(mta)s-whois[name=%(__name__)s, dest="%(destemail)s"]
>
> logpath  = /var/log/asterisk/messages
>
> maxretry = 3
>
> findtime  = 300
>
> bantime  = -1
>
>
>
>
>
> in filter.d
>
> asterisk.conf
>
> failregex = ^%(__prefix_line)s%(log_prefix)s Registration from '[^']*'
> failed for '(:¥d+)?' - (Wrong password|Username/auth name mismatch|No
> matching peer found|Not a local domain|Device does not match ACL|Peer is
> not supposed to register|ACL error ¥(permit/deny¥)|Not a local domain)$
>
> ^%(__prefix_line)s%(log_prefix)s Call from '[^']*'
> ¥(:¥d+¥) to extension '[^']*' rejected because extension not found in
> context
>
> ^%(__prefix_line)s%(log_prefix)s Host  failed to
> authenticate as '[^']*'$
>
> ^%(__prefix_line)s%(log_prefix)s No registration for peer
> '[^']*' ¥(from ¥)$
>
> ^%(__prefix_line)s%(log_prefix)s Host  failed MD5
> authentication for '[^']*' ¥([^)]+¥)$
>
> ^%(__prefix_line)s%(log_prefix)s Failed to authenticate
> (user|device) [^@]+@¥S*$
>
> ^%(__prefix_line)s%(log_prefix)s hacking attempt detected
> ''$
>
> ^%(__prefix_line)s%(log_prefix)s SecurityEvent="(FailedACL|
> InvalidAccountID|ChallengeResponseFailed|InvalidPassword)",EventTV="([¥
> d-]+|%(iso8601)s)",Severity="[¥w]+",Service="[¥w]+",
> EventVersion="¥d+",AccountID="(¥d*|)",SessionID=".+
> ",LocalAddress="IPV[46]/(UDP|TCP|WS)/[¥da-fA-F:.]+/¥d+",
> RemoteAddress="IPV[46]/(UDP|TCP|WS)//¥d+"(,Challenge="[¥w/]+")?(,
> ReceivedChallenge="¥w+")?(,Response="¥w+",ExpectedResponse="¥w*")?(,
> ReceivedHash="[¥da-f]+")?(,ACLName="¥w+")?$
>
> ^%(__prefix_line)s%(log_prefix)s "Rejecting unknown SIP
> connection from "$
>
> ^%(__prefix_line)s%(log_prefix)s Request (?:'[^']*' )?from
> '[^']*' failed for '(?::¥d+)?'¥s¥(callid: [^¥)]*¥) - (?:No matching
> endpoint found|Not match Endpoint(?: Contact)? ACL|(?:Failed|Error) to
> authenticate)¥s*$
>
>
>
> failregex = NOTICE.* .*: Registration from '.*' failed for '' -
> Wrong password
>
> NOTICE.* .*: Registration from '.*' failed for ':.*' -
> No matching peer found
>
> NOTICE.* .*: Registration from '.*' failed for '' - No
> matching peer found
>
> NOTICE.* .*: Registration from '.*' failed for '' -
> Username/auth name mismatch
>
> NOTICE.* .*: Registration from '.*' failed for '' -
> Device does not match ACL
>
> NOTICE.* .*: Registration from '.*' failed for '' - Peer
> is not supposed to register
>
> NOTICE.* .*: Registration from '.*' failed for '' - ACL
> error (permit/deny)
>
> NOTICE.* .*: Registration from '.*' failed for '' -
> Device does not match ACL
>
> NOTICE.*  failed to authenticate as '.*'$
>
> NOTICE.* .*: No registration for peer '.*' ¥(from ¥)
>
> NOTICE.* .*: Host  failed MD5 authentication for '.*'
> (.*)
>
> NOTICE.* .*: Failed to authenticate user .*@.*
>
> NOTICE.* .*: Sending fake auth rejection for device
> .*¥;tag=.*
>
> NOTICE.* .*: Registration from '¥".*¥".*' failed for ''
> - No matching peer found
>
> NOTICE.* .*: Registration from '¥".*¥".*' failed for ''
> - Wrong password
>
>
>
> ignoreregex =
>
>
>
> Thanks
>
> Motty
>
> --
> _
> -- Bandwidth and Colocation 

Re: [asterisk-users] BUG or ???

[Антон Сацкий]

Thanks U Richard
i know about this solution
but the main question why "${} substitution containing
the SHELL is evaluated before anything else"
Can U describe the rules   when and  why it happens?
Thanks

2017-02-24 23:44 GMT+02:00 Richard Mudgett <rmudg...@digium.com>:

>
>
> On Fri, Feb 24, 2017 at 3:30 PM, Антон Сацкий <satski...@gmail.com> wrote:
>
>> Got a strange situation
>>
>> [ext-queues]
>> ...
>> exten => h,2,ExecIf($[${CALLERID(num)} = ' ']?Set(var29=${SHELL(curl -X
>> POST --header "Content-Type: application/json" --header "Accept:
>> application/json" -d "{\"Phone\": ${FROMEXTEN}, \"Source\": \"asterisk\"}" "
>> http://sIte.com:80/api/v1/calls?apiKey=UABVAEI=3;)}))
>>
>> exten => h,3,NoOp(${var29})
>> exten => h,4,Macro(hangupcall,)
>> ;--== end of [ext-queues] ==--;
>>
>>
>> so when i execute  it got
>>
>>
>> -- Executing [h@ext-queues:1] NoOp("SIP/100-0050", "100") in new
>> stack
>> -- Executing [h@ext-queues:2] ExecIf("SIP/100-0050", "0
>> ?Set(var29=[{"RequestedCount":0,"MissedCount":7,"Total":7}])") in new
>> stack
>> -- Executing [h@ext-queues:3] NoOp("SIP/100-0050", "") in new
>> stack
>> -- Executing [h@ext-queues:4] Macro("SIP/100-0050",
>> "hangupcall,") in new st
>>
>>
>> U can see  that Execif = 0 = falce   but   somehow
>> Shell ${SHELL(curl -X POST --header "Content-Type: application/json"
>> --header "Accept: application/json" -d "{\"Phone\": ${FROMEXTEN},
>> \"Source\": \"asterisk\"}" "http://sIte.com:80/api/v1/cal
>> ls?apiKey=UABVAEI=3")}
>> executes and  get answer  from the server
>> [{"RequestedCount":0,"MissedCount":7,"Total":7}]
>>
>
> The Set isn't being executed by the ExecIf.  However the ${} substitution
> containing
> the SHELL is evaluated before anything else is examined.  This isn't a bug
> but the
> order of how things are evaluated.  You will have to do what you want
> another way:
>
> same = n,GotoIf($["${CALLERID(num)}" = ""]?skip)
> same = n,Set(var29=${SHELL(...)})
> same = n(skip),NoOp(${var29})
>
> Richard
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Best regards
Antony
tel.   +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy
<http://paypal.me/Satskiy?ppid=PPC000654=PL=en_PL(en_DK)=NN8XJS9XEP22C=21db79ac-ef8d-11e5-9553-9c8e992ea258==4d776c21ca7d2=4d776c21ca7d2=4d776c21ca7d2_tpcid=ppme-social-business-profile-created=main:email=main:email=op=em=ci=sys>
satski...@gmail.com <mail%3asatski...@gmail.com>
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[asterisk-users] BUG or ???

[Антон Сацкий]

Got a strange situation

[ext-queues]
...
exten => h,2,ExecIf($[${CALLERID(num)} = ' ']?Set(var29=${SHELL(curl -X
POST --header "Content-Type: application/json" --header "Accept:
application/json" -d "{\"Phone\": ${FROMEXTEN}, \"Source\": \"asterisk\"}" "
http://sIte.com:80/api/v1/calls?apiKey=UABVAEI=3;)}))

exten => h,3,NoOp(${var29})
exten => h,4,Macro(hangupcall,)
;--== end of [ext-queues] ==--;


so when i execute  it got


-- Executing [h@ext-queues:1] NoOp("SIP/100-0050", "100") in new stack
-- Executing [h@ext-queues:2] ExecIf("SIP/100-0050",
"0?Set(var29=[{"RequestedCount":0,"MissedCount":7,"Total":7}])")
in new stack
-- Executing [h@ext-queues:3] NoOp("SIP/100-0050", "") in new stack
-- Executing [h@ext-queues:4] Macro("SIP/100-0050", "hangupcall,")
in new st


U can see  that Execif = 0 = falce   but   somehow
Shell ${SHELL(curl -X POST --header "Content-Type: application/json"
--header "Accept: application/json" -d "{\"Phone\": ${FROMEXTEN},
\"Source\": \"asterisk\"}" "
http://sIte.com:80/api/v1/calls?apiKey=UABVAEI=3;)}
executes and  get answer  from the server
[{"RequestedCount":0,"MissedCount":7,"Total":7}]

i dont want it to be  executed



Thanks list for your help


-- 
Best regards
Antony
tel.   +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy

satski...@gmail.com 
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Re: [asterisk-users] Disallow CALLS without registry

[Антон Сацкий]

sorry
NoOP(${DB_EXISTS(SIP/Registry/${CHANNEL(peername)})});

2017-02-13 19:31 GMT+02:00 Антон Сацкий <satski...@gmail.com>:

> THINK i found a solution
>
> NoOP(${DB_EXISTS(/SIP/Registry/${CHANNEL(peername)})});
>
> THANKS TO ALL
>
> 2017-02-12 12:34 GMT+02:00 Frank Vanoni <mailingl...@linuxista.com>:
>
>> On Sat, 2017-02-11 at 12:25 +1300, Pete Mundy wrote:
>>
>> > > sip.conf configuration
>> > > In the [general] section, define:
>> > > [general]
>> > > ...
>> > > allowguest=no
>> > > alwaysauthreject=yes
>> > > ...
>>
>>
>> >
>>
>> With the above configuration on my Asterisk, I obtain the following
>> result:
>>
>> - if the phone is registered to Asterisk, I can place any call according
>> to the dial plan.
>>
>> - if the phone is NOT registered and I try to place a call, the phone
>> obtains a "403 forbidden" at any calling attempt.
>>
>>
>> Now, English is not my native language, but as far as I can understand,
>> "forbidden" means "not allowed" or "disallowed".
>>
>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Best regards
> Antony
> tel.   +380669197533 <+380%2066%20919%207533>
> tel2. +380636564340 <+380%2063%20656%204340>
> Paypal http://paypal.me/Satskiy
> <http://paypal.me/Satskiy?ppid=PPC000654=PL=en_PL(en_DK)=NN8XJS9XEP22C=21db79ac-ef8d-11e5-9553-9c8e992ea258==4d776c21ca7d2=4d776c21ca7d2=4d776c21ca7d2_tpcid=ppme-social-business-profile-created=main:email=main:email=op=em=ci=sys>
> satski...@gmail.com <mail%3asatski...@gmail.com>
>



-- 
Best regards
Antony
tel.   +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy
<http://paypal.me/Satskiy?ppid=PPC000654=PL=en_PL(en_DK)=NN8XJS9XEP22C=21db79ac-ef8d-11e5-9553-9c8e992ea258==4d776c21ca7d2=4d776c21ca7d2=4d776c21ca7d2_tpcid=ppme-social-business-profile-created=main:email=main:email=op=em=ci=sys>
satski...@gmail.com <mail%3asatski...@gmail.com>
-- 
_
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Re: [asterisk-users] Disallow CALLS without registry

[Антон Сацкий]

THINK i found a solution

NoOP(${DB_EXISTS(/SIP/Registry/${CHANNEL(peername)})});

THANKS TO ALL

2017-02-12 12:34 GMT+02:00 Frank Vanoni :

> On Sat, 2017-02-11 at 12:25 +1300, Pete Mundy wrote:
>
> > > sip.conf configuration
> > > In the [general] section, define:
> > > [general]
> > > ...
> > > allowguest=no
> > > alwaysauthreject=yes
> > > ...
>
>
> >
>
> With the above configuration on my Asterisk, I obtain the following
> result:
>
> - if the phone is registered to Asterisk, I can place any call according
> to the dial plan.
>
> - if the phone is NOT registered and I try to place a call, the phone
> obtains a "403 forbidden" at any calling attempt.
>
>
> Now, English is not my native language, but as far as I can understand,
> "forbidden" means "not allowed" or "disallowed".
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Best regards
Antony
tel.   +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy

satski...@gmail.com 
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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Re: [asterisk-users] Disallow CALLS without registry

[Антон Сацкий]

Thanks Frank -- but this not   a solution
below my  current  config

[general]

;sms
accept_outofcall_message= yes
outofcall_message_context   = messages
auth_message_requests   = no

;general
allowguest  = no
jbenable= no
jbimpl  = adaptive
allow   = !all,g722,ulaw,gsm
udpbindaddr = 0.0.0.0
transport   = udp

language= ru
context = public
alwaysauthreject= yes
nat = force_rport,comedia
directmedia = no
allowoverlap= no
match_auth_username = yes

progressinband  = yes
textsupport = yes
videosupport= yes
maxcallbitrate  = 1384
;
sendrpid = pai
rpid_update = yes
pedantic=no
 ;tos
tos_sip=cs3
tos_audio=ef
tos_video=cs4

2017-02-10 16:40 GMT+02:00 Frank Vanoni <mailingl...@linuxista.com>:

> On Thu, 2017-02-09 at 14:58 +0200, Антон Сацкий wrote:
>
>
> > so the main question is -- how to Disallow CALLS without registering
> > on PBX
>
> sip.conf configuration
> In the [general] section, define:
>
>
> [general]
> ...
> allowguest=no
> alwaysauthreject=yes
> ...
>
>
> The "allowguest" line disables anonymous SIP calls to your PBX. Some SIP
> providers connect as a guest user, however, so this may be inappropriate
> for your situation. Also, if you want to accept anonymous SIP calls,
> this line would block them, so you wouldn't want that. But it is listed
> here because it is the safest configuration.
>
> The "alwaysauthreject" line is important. This causes a hacker to get
> the same response from your PBX when they try to guess passwords whether
> or not they guessed a valid username. This also has the side-effect of
> making poorly written scanning scripts (the vast majority of hacker
> scripts seem to be poorly written) take less resources on your Asterisk
> box, as even if they scan a valid username, they'll think it doesn't
> exist.
>
> (Source: https://www.voip-info.org/wiki/view/Asterisk+security )
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
Best regards
Antony
tel.   +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy
<http://paypal.me/Satskiy?ppid=PPC000654=PL=en_PL(en_DK)=NN8XJS9XEP22C=21db79ac-ef8d-11e5-9553-9c8e992ea258==4d776c21ca7d2=4d776c21ca7d2=4d776c21ca7d2_tpcid=ppme-social-business-profile-created=main:email=main:email=op=em=ci=sys>
satski...@gmail.com <mail%3asatski...@gmail.com>
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[asterisk-users] Disallow CALLS without registry

[Антон Сацкий]

HI ALL
got  small question

i use call-limit=1 on peers


but  call limit is not working  if  user  is not  registered on PBX and
making calls

so the main question is -- how to Disallow CALLS without registering on PBX






-- 
Best regards
Antony
tel.   +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy

satski...@gmail.com 
-- 
_
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Re: [asterisk-users] Error when installing asterisk on Ubuntu 16.03LTS

[Антон Сацкий]

try to uncheck chan_unistim.so module

2016-10-08 14:57 GMT+03:00 christopher kamutumwa :

> Hello,
> I am trying to install asterisk 14 on ubuntu 16 but i am getting below
> error message please assist with how to resolve that after i run make
> && make install && make config && make samples
>
>[LD] astdb2bdb.o db1-ast/libdb1.a -> astdb2bdb
>[CPP] chan_unistim.c -> chan_unistim.i
>[CCi] chan_unistim.i -> chan_unistim.o
>[LD] chan_unistim.o -> chan_unistim.so
> /usr/bin/ld: /usr/lib/gcc/x86_64-linux-gnu/5/crtbeginT.o: relocation
> R_X86_64_32 against `__TMC_END__' can not be used when making a shared
> object; recompile with -fPIC
> /usr/lib/gcc/x86_64-linux-gnu/5/crtbeginT.o: error adding symbols: Bad
> value
> collect2: error: ld returned 1 exit status
> /usr/src/asterisk-14.0.2/Makefile.rules:176: recipe for target
> 'chan_unistim.so' failed
> make[1]: *** [chan_unistim.so] Error 1
> Makefile:397: recipe for target 'channels' failed
> make: *** [channels] Error 2
>
> thanks
>
> chri
>
> --
> _
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>
> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>   http://www.asterisk.org/community/astricon-user-conference
>
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>
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>



-- 
Best regards
Antony
tel.   +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy

satski...@gmail.com 
-- 
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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Антон Сацкий]

Try delete nat from 77wrtc settings ice should do the same

On Aug 11, 2016 10:00 PM, "Jonas Kellens"  wrote:

> On 11-08-16 18:03, Matt Fredrickson wrote:
>
>> On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens 
>> wrote:
>>
>>> My main reason not to upgrade to Ast 13 is because I'm afraid of losing
>>> functionality as there are certain functions deprecated/replaced. This
>>> can
>>> also cause headache :-)
>>>
>>> I will do so if there is no other option.
>>>
>>> But still, I don't see why Ast 13 would differ so much in this case ? If
>>> ICE
>>> and NAT is working (not causing problems) why should Ast 13 bring me
>>> audio
>>> and Ast 12 don't ??
>>>
>> If you want to minimize grief, start with 13 - WebRTC has been a
>> moving target for the last 5 years, it is not an old, mature standard
>> like ISDN or SIP.  If you find interop problems in an older version of
>> Asterisk with WebRTC, it's likely that it has been fixed in 13, and if
>> it hasn't the most likely place to obtain the fix will be in 13.
>>
>> After you get the WebRTC part working, then you can move back the
>> versions of Asterisk you're using to see if it still works.
>>
>> As far as ICE not working goes, if the browser you're talking to is
>> not on the same network as the Asterisk server, it's *possible* you
>> might need a true TURN server as well, instead of just an ICE server.
>>
>> Matthew Fredrickson
>>
>>
> Matthew
>
> when I set the following in rtp.conf :
>
> turnaddr=192.158.29.39:3478?transport=udp
> turnusername=28224511:1379330808
> turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA
>
>
> then Asterisk 12 gets really slow and sometimes unresponsive. Calls result
> in 480 request timeout (possibly due to the freeze of Asterisk).
>
> So this is also no solution.
>
> Can not even test if it brings me some audio in my webRTC calls.
>
>
> (putting the above lines back in comment resolves the issue of Asterisk
> freeze. This is all EXTREMELY BUGGY !)
>
>
> Asterisk 13 here I come (with very high expectations).
>
>
> Kind regards.
>
>
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[asterisk-users] need help

[Антон Сацкий]

please help me to solve the problem
if U can solve it for a chocolate :) it is also ok

https://issues.asterisk.org/jira/browse/ASTERISK-26073

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[asterisk-users] WebSocket connection from forcefully closed due to fatal write error

[Антон Сацкий]

Hi List
my CLI full of
WebSocket connection from "X"  forcefully closed due to fatal write
error
What should I do?

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[asterisk-users] WSS ISSUE

[Антон Сацкий]

HI all

can anybody help me there to search a problem
from time to time "Connection closed before receiving a handshake response"

WebSocket connection to 'wss://XXX:8089/ws' failed: Connection
closed before receiving a handshake response
sipml.js?14636613801642821:16763 ws_close
(index):2731 failed_to_start - Failed to connet to the server
sipml.js?14636613801642821:16764 WebSocket connection to
'wss://X:8089/ws' failed: Connection closed before receiving a
handshake response

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[asterisk-users] Strange SIP debug

[Антон Сацкий]

Hi list need your advice
i dont understand why reply ACK goes to diferrent ip (192.168.88.32)
SCREEN below

http://tinypic.com/view.php?pic=s6m7me=9#.VzsVhvl96Ik

THANK U ALL



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Re: [asterisk-users] Client TLS certificates for auth ?

[Антон Сацкий]

But what is the problem even if somehow your password will be stolen hacker
can't make a call because he needs certificate.of course if U setup ext to
use TLS only
On Mar 29, 2016 5:32 PM, "Markos Vakondios"  wrote:

> This would be very interesting, as we could register SIP devices securely
> over the internet without the need for VPN.
> Asterisk of course must accept only trusted client certificates the same
> way an OpenVPN server does.
> Anyone operating his/her remote endpoints like this?
> Anyone advising against this solution?
>
> On 29 March 2016 at 04:51, Kevin Long  wrote:
>
>>
>>
>> I use TLS and SRTP on my Asterisk servers. The server certificates are
>> signed by my internal CA, and the Root CA cert is distributed to the phones
>> and soft phones so they will trust the server without warning.
>>
>> It is not clear to me if Asterisk can be configured to actually reject
>> client connections/registrations from peers which do not possess a client
>> certificate which has been signed by a particular CA ?
>>
>> If so, could it be such that the common name in the client certificate
>> would need to match the username or Asterisk “extension” ?
>>
>>
>> I’m wondering if this can be done ,  to have a second factor of
>> authentication besides the SIP secret , since in my current setup, despite
>> using a TLS/SSL cert for the server, the server only verifies the client by
>> the SIP secret.
>>
>> Regards,
>>
>> Kevin Long
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>>
>
>
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Re: [asterisk-users] ODBC crashing asterisk

[Антон Сацкий]

You have an error in your SQL syntax; check the manual that corresponds
On Mar 23, 2016 11:38 PM, "Mike Diehl"  wrote:

> Hi all,
>
> I've got a new server up, but it's not staying up
>
> After a day or so, it segfaults with:
>
> [Mar 22 23:17:49] WARNING[12177]: res_odbc.c:1406 _ast_odbc_request_obj2:
> SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC
> 5.2(a)
> Driver]You have an error in your SQL syntax; check the manual that
> corresponds
> to your MySQL server version for the right syntax to use near '7' at line 1
>
>
> I'm using ODBC for sip and voice mail configuration.
>
> I'm running Asterisk 11.20.0-rc3.
>
> I've been told that there is a particular version of odbc that is stable.
> In
> the mean time, I'm trying to run unixODBC 2.3.2.
>
> What version SHOULD I use?
>
> TIA,
>
>
> --
> Mike Diehl
> Diehlnet Communications, LLC.
> Voice: (505) 903-5700
> Fax: (505) 903-5701
>
>
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[asterisk-users] WRONG Queues log

[Антон Сацкий]

Hi list need your help
i have call in queue it shows that it was answered by 4003

[root@asterisk ~]# grep --color "1456128646.157422"
/var/log/asterisk/queue_log-20160228

1456128688|1456128646.157422|800|NONE|ENTERQUEUE||0967145750|2
1456128717|1456128646.157422|800|SIP/4003|CONNECT|29|1456128688.157426|28
1456128817|1456128646.157422|800|SIP/4003|COMPLETECALLER|29|100|2


BUT IN FACT call was PICK UPPED  by 4001  using features

[root@asterisk ~]# grep --color "1456128646.157422"
/var/log/asterisk/full-20160228
[Feb 22 10:11:28] VERBOSE[9760][C-f165] pbx.c: -- Executing
[~~s~~@mix:2] MSet("SIP/3590640-000209b9",
"CDR(recordingfile)=3590640_1456128646.157422") in new stack
[Feb 22 10:11:28] VERBOSE[9760][C-f165] pbx.c: -- Executing
[~~s~~@mix:3] MixMonitor("SIP/3590640-000209b9",
"3590640_1456128646.157422.wav,b") in new stack



[root@asterisk ~]# grep --color "C-f165" /var/log/asterisk/full-20160228
[Feb 22 10:10:46] VERBOSE[2070][C-f165] netsock2.c:   == Using SIP RTP
CoS mark 5
[Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
[3590640@incoming:1] Set("SIP/3590640-000209b9", "CALLERID(name)=RU") in
new stack
[Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
[3590640@incoming:2] GotoIfTime("SIP/3590640-000209b9",
"9:00-19:30,mon-fri,*,*?4") in new stack
[Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Goto
(incoming,3590640,4)
[Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
[3590640@incoming:4] Goto("SIP/3590640-000209b9", "working") in new stack
[Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Goto
(incoming,3590640,13)
[Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
[3590640@incoming:13] Progress("SIP/3590640-000209b9", "") in new stack
[Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
[3590640@incoming:14] MSet("SIP/3590640-000209b9", "EXT=3590640") in new
stack
[Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
[3590640@incoming:15] Set("SIP/3590640-000209b9", "CHANNEL(language)=ru")
in new stack
[Feb 22 10:10:46] VERBOSE[9760][C-f165] pbx.c: -- Executing
[3590640@incoming:16] Playback("SIP/3590640-000209b9", "01_HELLO/01_HELLO")
in new stack
[Feb 22 10:10:46] VERBOSE[9760][C-f165] res_rtp_asterisk.c:>
0x7f9b1c19d490 -- Probation passed - setting RTP source address to
95.67.3.3:14380
[Feb 22 10:10:46] VERBOSE[9760][C-f165] file.c: --
 Playing '01_HELLO/01_HELLO.slin' (language 'ru')
[Feb 22 10:10:49] VERBOSE[9760][C-f165] pbx.c: -- Executing
[3590640@incoming:17] Wait("SIP/3590640-000209b9", "2") in new stack
[Feb 22 10:10:51] VERBOSE[9760][C-f165] pbx.c: -- Executing
[3590640@incoming:18] BackGround("SIP/3590640-000209b9",
"02_CHOICE_LANGUAGES/02_CHOICE_LANGUAGES") in new stack
[Feb 22 10:10:51] VERBOSE[9760][C-f165] file.c: --
 Playing
'02_CHOICE_LANGUAGES/02_CHOICE_LANGUAGES.slin' (language 'ru')
[Feb 22 10:10:55] DTMF[9760][C-f165] channel.c: DTMF begin '2' received
on SIP/3590640-000209b9
[Feb 22 10:10:55] DTMF[9760][C-f165] channel.c: DTMF begin ignored '2'
on SIP/3590640-000209b9
[Feb 22 10:10:55] DTMF[9760][C-f165] channel.c: DTMF end '2' received
on SIP/3590640-000209b9, duration 260 ms
[Feb 22 10:10:55] DTMF[9760][C-f165] channel.c: DTMF end passthrough
'2' on SIP/3590640-000209b9
[Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c:   == CDR updated on
SIP/3590640-000209b9
[Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Executing
[2@incoming:1] Set("SIP/3590640-000209b9", "CHANNEL(language)=ua") in new
stack
[Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Executing
[2@incoming:2] Set("SIP/3590640-000209b9", "CALLERID(name)=UA") in new stack
[Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Executing
[2@incoming:3] Goto("SIP/3590640-000209b9", "ua_start,3590640,1") in new
stack
[Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Goto
(ua_start,3590640,1)
[Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Executing
[3590640@ua_start:1] Set("SIP/3590640-000209b9", "CHANNEL(language)=ua") in
new stack
[Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Executing
[3590640@ua_start:2] Set("SIP/3590640-000209b9", "TIMEOUT(digit)=3") in new
stack
[Feb 22 10:11:00] VERBOSE[9760][C-f165] func_timeout.c: -- Digit
timeout set to 3.000
[Feb 22 10:11:00] VERBOSE[9760][C-f165] pbx.c: -- Executing
[3590640@ua_start:3] BackGround("SIP/3590640-000209b9",
"01_QUALITY_OF_THE_SERVICE/01_QUALITY_OF_THE_SERVICE") in new stack
[Feb 22 10:11:00] VERBOSE[9760][C-f165] file.c: --
 Playing
'01_QUALITY_OF_THE_SERVICE/01_QUALITY_OF_THE_SERVICE.slin' (language 'ua')
[Feb 22 10:11:22] DTMF[9760][C-f165] channel.c: DTMF begin '3' received
on SIP/3590640-000209b9
[Feb 22 10:11:22] DTMF[9760][C-f165] channel.c: DTMF begin ignored '3'
on SIP/3590640-000209b9
[Feb 22 

Re: [asterisk-users] Asterisk Manager Interface AMI over HTTP.

[Антон Сацкий]

Check http.conf
On Aug 31, 2015 5:31 PM, "Aziz TestAccount"  wrote:

> Hi All!
>
> I'm using Asterisk 11.6-cert11 and trying to set the AMI over HTTP ,
> without success. I always get the Error :
>
> ---
>
> Asterisk Call Manager/1.3
> Response: Error
> Message: Missing action in request
> ---
>
>
> I made the following configuration in manager.conf :
>
> [general]
> enabled = yes
> webenabled = yes
> enablestatic=yes
> port = 
> bindaddr = 0.0.0.0
>
> [admin]
> secret = admin1234
> deny=0.0.0.0/0.0.0.0
> permit=192.168.1.0/255.255.255.0 
> 
> read = system,call,log,verbose,command,agent,user,originate
> write = system,call,log,verbose,command,agent,user,originate
>
>
> And I'm trying to access via HTTP using the following link : 
> http://192.168.1.134:/manager?action=login=aziz=aziz1234/
>
>
> Could anyone tell me if I'm missing something.
>
> Thanks in advance.
>
>
>
>
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Re: [asterisk-users] What database should I use, for simple data storing? SQLite or the buitin one?

[Антон Сацкий]

Propose U to use Mysql

2015-07-07 17:26 GMT+03:00 Rodrigo Pimenta Carvalho pime...@inatel.br:



 Hi.

 I was studying about how to use databases in Asterisk, accessing it from
 the dial plan.
 In my project, my dial plan will have to store simple data (ex: IP number,
 port number, device name, etc) in a persistent way, so that it will be
 possible to retrieve such information in future moments, still via dial
 plan.

 For this case, I would like to know?

 1. What is the best choice for storing and retrieving simple data , with
 dial plan instructions: SQLite or the builtin database option? Consider
 that I'm worried about installation, configuration and use difficulties.

 2. Does Asterisk 13 come with SQLite ready for use or have I to install
 this database separately and configure it to be accessible in dial plan?

 3. Where can I find tutorials about using SQLite or the builtin database
 for storing simple that?

 P.S.: I'm not interested in storing CDR data.

 Any hint will be very helpful!

 Thanks a lot!


 RODRIGO PIMENTA CARVALHO
 Inatel Competence Center
 Software
 Ph: +55 35 3471 9200 RAMAL 979(Brasil)
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Antony
моб (066) 919-75-33
моб (063) 656-43-40
satski...@gmail.com mail%3asatski...@gmail.com
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[asterisk-users] Unisteam not showing callerid

[Антон Сацкий]

hi list
can U help me

caller id in USTM if now working



  -- Starting switch on '4211@4211-1' to 4203
-- Executing [4203@office:1] DumpChan(USTM/4211@4211-0x7f7ba4228fd0,
) in new stack

Dumping Info For Channel: USTM/4211@4211-0x7f7ba4228fd0:

Info:
Name=   USTM/4211@4211-0x7f7ba4228fd0
Type=   USTM
UniqueID=   1436177628.8051
LinkedID=   1436177628.8051
CallerIDNum=(N/A)
CallerIDName=   (N/A)
ConnectedLineIDNum= (N/A)
ConnectedLineIDName=(N/A)
DNIDDigits= (N/A)
RDNIS=  (N/A)
Parkinglot=
Language=   en
State=  Ring (4)
Rings=  0
NativeFormat=   (ulaw|alaw)
WriteFormat=ulaw
ReadFormat= ulaw
RawWriteFormat= ulaw
RawReadFormat=  ulaw
WriteTranscode= No
ReadTranscode=  No
1stFileDescriptor=  1652
Framesin=   0
Framesout=  0
TimetoHangup=   0
ElapsedTime=0h0m0s
BridgeID=   (Not bridged)
Context=office
Extension=  4203
Priority=   1
CallGroup=  4
PickupGroup=4
Application=DumpChan
Data=   (Empty)
Blocking_in=(Not Blocking)

Variables:

-- Executing [4203@office:2] Dial(USTM/4211@4211-0x7f7ba4228fd0,
USTM/4203@4203) in new stack







my CONF



general]
port=5000





 [unistim-phones](!)
 bookmark=Support@123;Softkey to speed dial
 context=office
; extension=line

  rtp_method=1

 maintext0=GREETING  ; default = Welcome, 24 characters max
 maintext1=have a great day   ; default = the name of the device, 24
characters max
 dateformat=1; 0 = month/day, 1 (default) = day/month
 timeformat=2; 0 = 0:00am ; 1 (default) = 0h00, 2 = 0:00
 callhistory=0   ; 0 = disable, 1 = enable call history,
default = 1


[group3]
callgroup   = 3
pickupgroup = 3

[group4]
callgroup   = 4
pickupgroup = 4

[group5]
callgroup   = 3,5
pickupgroup = 3,5

[group6]
callgroup   = 6
pickupgroup = 6



[4294](unistim-phones,group3)
device=0016caf460f5
line= 4294
callerid=Victoriya Mukan 4294

[4211](unistim-phones,group4)
device=000ae475faed
line= 4211
callerid=Gomenyuk tatyana 4211


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Antony
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моб (063) 656-43-40
satski...@gmail.com mail%3asatski...@gmail.com
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[asterisk-users] multiple sip trunks with the same ITSP

[Антон Сацкий]

HI LIST CAN U HELP ME

If there are multiple sip trunks with the same ITSP then an incoming call
is arbitarily matched to the last peer with the same host IP address. This
is not a serious problem because the DID is still correct but it does have
many insidious effects due to the incorrect channel name

Example

register=myaccou...@sip.myitsp.com/line1
register=myaccou...@sip.myitsp.com/line2
[line1]
type=peer
username=myaccount1
host=sip.myitsp.com
[line2]
type=peer
username=myaccount2
host=sip.myitsp.com

If sip.myitsp.com directs a call to asterisk with a request line of:

INVITE line1@mybindaddr SIP/2.0

then it is matched to the line2 peer whereas it would probably be better
matched to the line1 peer


-- 
Best regards
Antony
моб (066) 919-75-33
моб (063) 656-43-40
satski...@gmail.com mail%3asatski...@gmail.com
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_
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[asterisk-users] hi list need your help

[Антон Сацкий]

facing problem with  originating  webrtc calls


1-when iam  doing call from webrtc iget ice working
--- SIP read from WS:91.196.158.205:1466 ---
INVITE sip:0669197533@77.91.132.9 SIP/2.0
Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315
Max-Forwards: 69
To: sip:0669197533@77.91.132.9
From: Anton sip:1065@77.91.132.9;tag=5i21qaop43
Call-ID: ocq4hu8eol3kijsgvt6b
CSeq: 1465 INVITE
Authorization: Digest algorithm=MD5, username=1065, realm=77.91.132.9,
nonce=5152b137, uri=sip:0669197533@77.91.132.9,
response=446883f3c97a49ea7a9a554a1ba31b6a
X-Can-Renegotiate: true
Contact: sip:0momhddj@7cvtd9ihs2e8.invalid;transport=ws;ob
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.26
Content-Length: 2554

v=0
o=- 4785391175048354014 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br
m=audio 2313 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 192.168.88.26
a=rtcp:2313 IN IP4 192.168.88.26
a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host
generation 0
a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host
generation 0
a=candidate:1263319685 1 tcp 1518280447 192.168.88.26 0 typ host tcptype
active generation 0
a=candidate:1263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype
active generation 0
a=ice-ufrag:8nMZ7w8bHdBBoY1a
a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR
a=fingerprint:sha-256
6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3696151487 cname:jXfPZ33h32Mx9liw
a=ssrc:3696151487 msid:cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br
8a2acec3-8511-4d36-9b51-05b8752c2ddd
a=ssrc:3696151487 mslabel:cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br
a=ssrc:3696151487 label:8a2acec3-8511-4d36-9b51-05b8752c2ddd
m=video 2313 RTP/SAVPF 100 116 117 96
c=IN IP4 192.168.88.26
a=rtcp:2313 IN IP4 192.168.88.26
a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host
generation 0
a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host
generation 0
a=candidate:1263319685 1 tcp 1518280447 192.168.88.26 0 typ host tcptype
active generation 0
a=candidate:1263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype
active generation 0
a=ice-ufrag:8nMZ7w8bHdBBoY1a
a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR
a=fingerprint:sha-256
6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=recvonly
a=rtcp-mux
a=rtpmap:100 VP8/9
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/9
a=rtpmap:117 ulpfec/9
a=rtpmap:96 rtx/9
a=fmtp:96 apt=100

2-BUT when i do channel originate sip/GOROD/X extension 1065@office
-- Executing [1065@office:1] Dial(SIP/GOROD-0004, SIP/1065) in
new stack
  == Using SIP RTP CoS mark 5
[Apr 28 14:07:47] ERROR[4006][C-0032]: netsock2.c:269
ast_sockaddr_resolve: getaddrinfo(7cvtd9ihs2e8.invalid, (null), ...):
Name or service not known
[Apr 28 14:07:47] WARNING[4006][C-0032]: chan_sip.c:15869
__set_address_from_contact: Invalid host name in Contact: (can't resolve in
DNS) : '7cvtd9ihs2e8.invalid'
[Apr 28 14:07:47] ERROR[4006][C-0032]: netsock2.c:98
ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
Audio is at 16476
Adding codec 13 (ulaw) to SDP
Adding codec 12 (gsm) to SDP
Adding codec 14 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 91.196.158.205:1466:
INVITE sip:0momhddj@7cvtd9ihs2e8.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 77.91.132.9:5060;branch=z9hG4bK3f293d79;rport
Max-Forwards: 70
From: asterisk sip:asterisk@77.91.132.9;tag=as78119d2b
To: sip:0momhddj@7cvtd9ihs2e8.invalid;transport=ws
Contact: sip:asterisk@77.91.132.9:5060;transport=WS
Call-ID: 17a96e0848cdd7d226d3665a36c65c77@77.91.132.9:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.15.0
Date: Tue, 28 Apr 2015 11:07:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 437

v=0
o=root 1122885298 1122885298 IN IP4 77.91.132.9
s=Asterisk PBX 11.15.0
c=IN IP4 77.91.132.9
t=0 0
m=audio 16476 RTP/SAVPF 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 

Re: [asterisk-users] Asterisk 11.17.0 Now Available

[Антон Сацкий]

Hi list can i ask U
does this release solved my problem
https://github.com/versatica/JsSIP/issues/311
(already try on a last FREEPBX --same issue)
Regards

2015-04-01 22:01 GMT+03:00 Asterisk Development Team 
asteriskt...@digium.com:

 The Asterisk Development Team has announced the release of Asterisk
 11.17.0.
 This release is available for immediate download at
 http://downloads.asterisk.org/pub/telephony/asterisk

 The release of Asterisk 11.17.0 resolves several issues reported by the
 community and would have not been possible without your participation.
 Thank you!

 The following are the issues resolved in this release:

 New Features made in this release:
 ---
  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
   (Reported by Dwayne Hubbard)

 Bugs fixed in this release:
 ---
  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
   res_odbc (Reported by ibercom)
  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
   with replaces (Reported by Eelco Brolman)
  * ASTERISK-24479 - Enable REF_DEBUG for module references
   (Reported by Corey Farrell)
  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
   fully disconnect underlying socket, leading to events being
   dropped with no additional information (Reported by Matt Jordan)
  * ASTERISK-24772 - ODBC error in realtime sippeers when device
   unregisters under MariaDB (Reported by Richard Miller)
  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
   (Reported by Corey Farrell)
  * ASTERISK-24799 - [patch] make fails with undefined reference to
   SSLv3_client_method (Reported by Alexander Traud)
  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
   for playing back messages stored in IMAP - play_message: No
   origtime (Reported by Graham Barnett)
  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
   OSX with 64 bit integers (Reported by Corey Farrell)
  * ASTERISK-24796 - Codecs and bucket schema's prevent module
   unload (Reported by Corey Farrell)
  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
   (Reported by Ashley Sanders)
  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
   (Reported by Kevin Harwell)
  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
   thread ID being passed to pthread_kill (Reported by JoshE)
  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
   fail (Reported by Terry Wilson)
  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
   SRTP for audio, but they responded without it' is ambiguous and
   wrong in some cases (Reported by Rusty Newton)
  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
   error response and BYE are sent to the caller (Reported by
   Makoto Dei)
  * ASTERISK-18105 - most of asterisk modules are unbuildable in
   cygwin environment (Reported by feyfre)
  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
   building a peer causes a peer poke during request handling
   (Reported by Richard Mudgett)
  * ASTERISK-24825 - Caller ID not recognized using
   Centrex/Distinctive dialing (Reported by Richard Mudgett)
  * ASTERISK-24739 - [patch] - Out of files -- call fails --
   numerous files with inodes from under /usr/share/zoneinfo,
   mostly posixrules (Reported by Ed Hynan)
  * ASTERISK-23390 - NewExten Event with application AGI shows up
   before and after AGI runs (Reported by Benjamin Keith Ford)
  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
   voicemail stored in LDAP (Reported by Graham Barnett)
  * ASTERISK-24808 - res_config_odbc: Improper escaping of
   backslashes occurs with MySQL (Reported by Javier Acosta)
  * ASTERISK-20850 - [patch]Nested functions aren't portable.
   Adapting RAII_VAR to use clang/llvm blocks to get the
   same/similar functionality. (Reported by Diederik de Groot)
  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
   by Frank DiGennaro)
  * ASTERISK-21038 - Bad command completion of core set debug
   channel (Reported by Richard Kenner)
  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
   Dave Cabot)
  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
   Atis Lezdins)
  * ASTERISK-24876 - Investigate reference leaks from
   tests/channels/local/local_optimize_away (Reported by Corey
   Farrell)
  * ASTERISK-24817 - init_logger_chain: unreachable code block
   (Reported by Corey Farrell)
  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
   snuffy)
  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
   under OpenBSD (Reported by snuffy)

 Improvements made in this release:
 

[asterisk-users] cant get incoming calls in asterisk

[Антон Сацкий]

*friends help me *
*cant get incoming calls in asterisk*
*(when i connect **80081 in softphone ---every thing is ok**)*


*--- SIP read from UDP:200.152.125.221:5060 http://200.152.125.221:5060
---*
*INVITE sip:80081@10.47.10.10:5060 http://sip:80081@10.47.10.10:5060
SIP/2.0*
*Record-Route: sip:200.152.125.221;lr;ftag=as6872d065*
*Via: SIP/2.0/UDP 200.152.125.221;branch=z9hG4bKd4fd.b3489837.0*
*Via: SIP/2.0/UDP 200.152.125.213:5060;branch=z9hG4bK2214551d;rport=5060*
*From: 8008  21982008200 sip:111...@ser.sipcode.com.br
sip%3a111...@ser.sipcode.com.br;tag=as6872d065*
*To: sip:80...@ser.sipcode.com.br sip%3a80...@ser.sipcode.com.br*
*Contact: sip:11@200.152.125.213 sip%3A11@200.152.125.213*
*Call-ID: 5c385d117f894c0f2dd79a3f2129b...@ser.sipcode.com.br
5c385d117f894c0f2dd79a3f2129b...@ser.sipcode.com.br*
*CSeq: 105 INVITE*
*User-Agent: FPBX-2.9.0(1.4.41)*
*Max-Forwards: 69*
*Date: Fri, 06 Mar 2015 18:17:21 GMT*
*Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO*
*Supported: replaces*
*Content-Type: application/sdp*
*Content-Length: 338*

*v=0*
*o=root 3211 3214 IN IP4 200.152.125.213*
*s=session*
*c=IN IP4 200.152.125.213*
*t=0 0*
*m=audio 14686 RTP/AVP 0 8 3 18 101*
*a=rtpmap:0 PCMU/8000*
*a=rtpmap:8 PCMA/8000*
*a=rtpmap:3 GSM/8000*
*a=rtpmap:18 G729/8000*
*a=fmtp:18 annexb=no*
*a=rtpmap:101 telephone-event/8000*
*a=fmtp:101 0-16*
*a=silenceSupp:off - - - -*
*a=ptime:20*
*a=sendrecv*
*-*
*--- (16 headers 16 lines) ---*
*Sending to 200.152.125.221:5060 http://200.152.125.221:5060 (no NAT)*
*Using INVITE request as basis request -
5c385d117f894c0f2dd79a3f2129b...@ser.sipcode.com.br
5c385d117f894c0f2dd79a3f2129b...@ser.sipcode.com.br*
*Found peer '11' for '11' from 200.152.125.221:5060
http://200.152.125.221:5060*

*--- Reliably Transmitting (no NAT) to 200.152.125.221:5060
http://200.152.125.221:5060 ---*
*SIP/2.0 401 Unauthorized*
*Via: SIP/2.0/UDP
200.152.125.221;branch=z9hG4bKd4fd.b3489837.0;received=200.152.125.221*
*Via: SIP/2.0/UDP 200.152.125.213:5060;branch=z9hG4bK2214551d;rport=5060*
*From: 8008  21982008200 sip:111...@ser.sipcode.com.br
sip%3a111...@ser.sipcode.com.br;tag=as6872d065*
*To: sip:80...@ser.sipcode.com.br
sip%3a80...@ser.sipcode.com.br;tag=as09849411*
*Call-ID: 5c385d117f894c0f2dd79a3f2129b...@ser.sipcode.com.br
5c385d117f894c0f2dd79a3f2129b...@ser.sipcode.com.br*
*CSeq: 105 INVITE*
*Server: FPBX-12.0.42(11.14.1)*
*Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE*
*Supported: replaces, timer*
*WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=63fdf36b*
*Content-Length: 0*


**






-- 
Best regards
Antony
моб (066) 919-75-33
моб (063) 656-43-40
satski...@gmail.com mail%3asatski...@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

Re: [asterisk-users] Callfile problem - Unable to find codec translation path from (nothing)

[Антон Сацкий]

1-wrong AAA/check_ip_failure--- try to use default sounds
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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