Re: [asterisk-users] Asterisk not using common codec between (SIP) endpoints

[Administrator TOOTAI]

Hi

Le 03/10/2019 à 13:13, Andreas Wehrmann a écrit :

[...]


- Even if direct_media is disabled: Is there a way to make Asterisk 
always use a common codec between SIP endpoints,

   so it doesn't need to transcode?


Before calling the gatreway add

same = n,set(SIP_CODEC=alaw)

[...]

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Re: [asterisk-users] Increasing variables - Changes v13 vs v16

[Administrator TOOTAI]

Le 01/10/2019 à 16:38, Eric Wieling a écrit :

Verify ${myCpt} is not empty.


Yes, it was that. Many thanks



On 10/1/19 10:24 AM, Administrator TOOTAI wrote:

Hi list,

on asterisk 13 I use

same => n,Set(__myCpt=$[${myCpt} + 1])

which is working well. On an Asterisk 16 I get, for this same command

[2019-10-01 16:15:01] WARNING[28197][C-0008]: ast_expr2.fl:470 
ast_yyerror: ast_yyerror():  syntax error: syntax error, unexpected 
'+', expecting $end; Input:


  + 1

  ^

What changes in 16 version creates this behavior and how to get it work ?

Thanks for any hint





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[asterisk-users] Increasing variables - Changes v13 vs v16

[Administrator TOOTAI]

Hi list,

on asterisk 13 I use

same => n,Set(__myCpt=$[${myCpt} + 1])

which is working well. On an Asterisk 16 I get, for this same command

[2019-10-01 16:15:01] WARNING[28197][C-0008]: ast_expr2.fl:470 
ast_yyerror: ast_yyerror():  syntax error: syntax error, unexpected '+', 
expecting $end; Input: 



 + 1 



 ^

What changes in 16 version creates this behavior and how to get it work ?

Thanks for any hint

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Re: [asterisk-users] Security AccountID unknown - PJSIP

[Administrator TOOTAI]

Le 30/09/2019 à 15:58, Joshua C. Colp a écrit :

On Mon, Sep 30, 2019, at 10:52 AM, Administrator TOOTAI wrote:

Le 30/09/2019 à 11:45, Joshua C. Colp a écrit :

On Fri, Sep 27, 2019, at 11:31 AM, Administrator TOOTAI wrote:

Hi list,

I would like to now what is the sense of such type of entry in security.log

[2019-09-27 15:12:24] SECURITY[26964] res_security_log.c:
SecurityEvent="ChallengeSent",EventTV="2019-09-27T15:12:24.181+0200",Severity="Informational",Servic
e="PJSIP",EventVersion="1",AccountID="",
SessionID="56b0ca9-d967a90d16411209-a1b0fae1@188.165.222.17",LocalAddress="IPV4/UDP//5060",
RemoteAddress="IPV4/UDP//5213",Challenge=""

We have a lot of such tries coming from IPs not allowed and fail2ban
fail to ban them because of SecurityEvent not treated and Severity
Informational.

We add a fail2ban filter to ban those IPs which is OK on our side but
also means that attacker knows that account is not existing.

Any comment appreciate


SIP uses a challenge/response mechanism for authentication. The above indicates 
that a challenge was sent. The remote side is under no obligation to retry with 
authentication and may choose not to. If they did and failed another message 
would occur.


  From security logs 26/09/2019 before we add our fail2ban rule:

EventTV="2019-09-26T14:32:06.516+0200",RemoteAddress="IPV4/UDP/66.117.9.138/52488"

EventTV="2019-09-26T14:32:45.748+0200",RemoteAddress="IPV4/UDP/66.117.9.138/57808"

EventTV="2019-09-26T14:33:25.300+0200",RemoteAddress="IPV4/UDP/66.117.9.138/63211"

EventTV="2019-09-26T14:34:04.527+0200",RemoteAddress="IPV4/UDP/66.117.9.138/51988"

In 2 minutes, the same IP address. We count 28862 tries from 11/09/2019
to 26/09/2019 coming *ONLY* from this IP address :(, average being 80
tries/hours.

If I understand you, there is no check between 2 authentication tries
coming from the same IP address which doesn't reply to a challenge ?


There is not. Asterisk doesn't keep track of such things.



OK, thanks
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Re: [asterisk-users] Security AccountID unknown - PJSIP

[Administrator TOOTAI]

Le 30/09/2019 à 11:45, Joshua C. Colp a écrit :

On Fri, Sep 27, 2019, at 11:31 AM, Administrator TOOTAI wrote:

Hi list,

I would like to now what is the sense of such type of entry in security.log

[2019-09-27 15:12:24] SECURITY[26964] res_security_log.c:
SecurityEvent="ChallengeSent",EventTV="2019-09-27T15:12:24.181+0200",Severity="Informational",Servic
e="PJSIP",EventVersion="1",AccountID="",
SessionID="56b0ca9-d967a90d16411209-a1b0fae1@188.165.222.17",LocalAddress="IPV4/UDP//5060",
RemoteAddress="IPV4/UDP//5213",Challenge=""

We have a lot of such tries coming from IPs not allowed and fail2ban
fail to ban them because of SecurityEvent not treated and Severity
Informational.

We add a fail2ban filter to ban those IPs which is OK on our side but
also means that attacker knows that account is not existing.

Any comment appreciate


SIP uses a challenge/response mechanism for authentication. The above indicates 
that a challenge was sent. The remote side is under no obligation to retry with 
authentication and may choose not to. If they did and failed another message 
would occur.


From security logs 26/09/2019 before we add our fail2ban rule:

EventTV="2019-09-26T14:32:06.516+0200",RemoteAddress="IPV4/UDP/66.117.9.138/52488" 

EventTV="2019-09-26T14:32:45.748+0200",RemoteAddress="IPV4/UDP/66.117.9.138/57808" 

EventTV="2019-09-26T14:33:25.300+0200",RemoteAddress="IPV4/UDP/66.117.9.138/63211" 


EventTV="2019-09-26T14:34:04.527+0200",RemoteAddress="IPV4/UDP/66.117.9.138/51988"

In 2 minutes, the same IP address. We count 28862 tries from 11/09/2019 
to 26/09/2019 coming *ONLY* from this IP address :(, average being 80 
tries/hours.


If I understand you, there is no check between 2 authentication tries 
coming from the same IP address which doesn't reply to a challenge ?


Thanks for your support

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[asterisk-users] Security AccountID unknown - PJSIP

[Administrator TOOTAI]

Hi list,

I would like to now what is the sense of such type of entry in security.log

[2019-09-27 15:12:24] SECURITY[26964] res_security_log.c: 
SecurityEvent="ChallengeSent",EventTV="2019-09-27T15:12:24.181+0200",Severity="Informational",Servic
e="PJSIP",EventVersion="1",AccountID="", 
SessionID="56b0ca9-d967a90d16411209-a1b0fae1@188.165.222.17",LocalAddress="IPV4/UDP//5060",

RemoteAddress="IPV4/UDP//5213",Challenge=""

We have a lot of such tries coming from IPs not allowed and fail2ban 
fail to ban them because of SecurityEvent not treated and Severity 
Informational.


We add a fail2ban filter to ban those IPs which is OK on our side but 
also means that attacker knows that account is not existing.


Any comment appreciate

Best Regards

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Re: [asterisk-users] if statement with true value that contains a colon

[Administrator TOOTAI]

Le 13/09/2019 à 14:03, Brian J. Murrell a écrit :

How can I use an IF statement with a true value being a variable that
has a colon in it?  The colon in the true value variable is being taken
as the delimiter for the false value.

The only solution I came up with was some hackery to use STRREPLACE to
replace the : with a % before the IF statement and then use STRREPLACE
again after to change the % back to a :.

i.e.:

 Set(dialexts=${STRREPLACE(dialexts,:,%)});
 Set(dialexts=${IFTIME(8:00-22:00?${dialexts}&${MBDR}:${dialexts})});
 Set(dialexts=${STRREPLACE(dialexts,%,:)});


Is there no better alternative?


Escape it with \

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Re: [asterisk-users] PJSIP reInvite

[Administrator TOOTAI]

Le 15/08/2019 à 14:06, Jöran Vinzens a écrit :

Hi,

we tried "direct_media=no". this is documented to suppress reInvites but 
it has no effect.
"directmedia" is not known by the config parser and it gives error while 
reading.


Speeking about directmedia was not to point you to a command ;) more to 
a general approach




direct_media=no is not the same behavior as canreinvite=no, at least as 
far I can see it.


Did you try direct_media_glare_mitigation ? See

https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_res_pjsip#Asterisk16Configuration_res_pjsip-endpoint_direct_media_method



BR
Jöran

On Thu, Aug 15, 2019 at 2:03 PM Administrator TOOTAI <mailto:ad...@tootai.net>> wrote:


Le 15/08/2019 à 13:22, Jöran Vinzens a écrit :
 > Hi All,
 >
 > We are using asterisk 16.5 and having an issue with the first
re-invite
 > after the call has been established.
 > We can see the call gets up and you see in the logs the bridge
type has
 > changed and after that a re-invite is triggered.
 >
 > Is there any possibility to deactivate this kind of reInvite? We
have
 > some race conditions while have multiple asterisk in the call
flow and
 > the different asterisk systems are sending this reInvites out
parallel.
 > While an invite is pending on a system it is not accepting another
 > incoming reInvite from peer.
 >
 > With chan_SIP canreinvite=no solved the issue. But it seems there is
 > nothing similar in PJSIP.

As far as I know directmedia is the replacement of canreinvite

[...]

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--

Jöran Vinzens -vinz...@sipgate.de  <mailto:vinz...@sipgate.de>
Telefon: +49 211-63 55 56-21
Telefax: +49 211-63 55 55-22

sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391

www.sipgate.de  <http://www.sipgate.de>  -www.sipgate.co.uk  
<http://www.sipgate.co.uk>




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Re: [asterisk-users] PJSIP reInvite

[Administrator TOOTAI]

Le 15/08/2019 à 13:22, Jöran Vinzens a écrit :

Hi All,

We are using asterisk 16.5 and having an issue with the first re-invite 
after the call has been established.
We can see the call gets up and you see in the logs the bridge type has 
changed and after that a re-invite is triggered.


Is there any possibility to deactivate this kind of reInvite? We have 
some race conditions while have multiple asterisk in the call flow and 
the different asterisk systems are sending this reInvites out parallel. 
While an invite is pending on a system it is not accepting another 
incoming reInvite from peer.


With chan_SIP canreinvite=no solved the issue. But it seems there is 
nothing similar in PJSIP.


As far as I know directmedia is the replacement of canreinvite

[...]

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[asterisk-users] Calls and queue statistics

[Administrator TOOTAI]

Hello list,

I'm looking for a solution that can be applied to a stock asterisk 16 
(pjsip if it matter) running Debian 9 (php7.0).


Statistics should be available for normal calls and queues using a WEB 
interface. Open source better but not necessary,


Any feedback appreciate, no matter if it's a "go for it" or "go away".

Regards

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Re: [asterisk-users] Dial(${PJSIP_DIAL_CONTACTS(Alice)} & ${PJSIP_DIAL_CONTACTS(Bob)}) how not to fail if one endpoint has no registered AOR?

[Administrator TOOTAI]

Le 10/06/2019 à 10:53, Benoit Panizzon a écrit :

What about to put eveything in a variable and the remove the last
character if it equal &


Yes, I considered this...

What if you dial three endpoints and the middle one (or last one) is
empty? You would also need to remove the first & and any double &
within that string. Is it faisable with asterisk logic?


When I'm in your case I add *not empty* EP one by one adding & after 
each. At the end, my variable is empty or finish with &. I then have to 
remove this last character and start dial.


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Re: [asterisk-users] Dial(${PJSIP_DIAL_CONTACTS(Alice)} & ${PJSIP_DIAL_CONTACTS(Bob)}) how not to fail if one endpoint has no registered AOR?

[Administrator TOOTAI]

Le 09/06/2019 à 13:19, Benoit Panizzon a écrit :

Dear List


Hello



It's probably been more than a year now I switched from chan_sip to
pjsip. pjsip works much cleaner than chan_sip.

But!

I have come across a Problem I was not able to solve with Asterisk
Dialplan Logic.

With pjsip an endpoint can have multiple AOR, so you need to expand
them with ${PJSIP_DIAL_CONTACTS()} to be able to Dial() all of them
simultaneously.

But there are also situation where you need to Dial() not only one
endpoint, but multiple ones, even mixing technologies like IAX and SIP.

You can specify those multiple endpoints with the & separator in the
Dial() function.

Unfortunately if an pjsip endpoint has NO registered AORs,
${PJSIP_DIAL_CONTACTS()} returns an empty sting.

So consider:

same => n,Dial(IAX2/gu...@pbx.digium.com/s@default &
${PJSIP_DIAL_CONTACTS(Guest)})

If there is no Guest registered, the resulting string to dial passed to
Dial() is: "IAX2/gu...@pbx.digium.com/s@default &" which Dial complains
is not valid, because of a missing second line to dial after the &.


What about to put eveything in a variable and the remove the last 
character if it equal &


Something like

same = 
n,Set(toDial=IAX2/gu...@pbx.digium.com/s@default&${PJSIP_DIAL_CONTACTS(Guest)})

same = n,ExecIf($["${toDial:-1}"=="&"]?Set(toDial=${toDial:0:-1}))
same = n,Dial(${toDial})

[...]
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Re: [asterisk-users] Fail2ban for asterisk 16 PJSIP

[Administrator TOOTAI]

Le 08/06/2019 à 05:20, John T. Bittner a écrit :

Hopefully, this helps someone else.


This seems to be working for me.

# Fail2Ban configuration file

[INCLUDES]

#before = common.conf

[Definition]

failregex = NOTICE.* .*: Request \'REGISTER\' from '.*' failed for 
':.*' .* - No matching endpoint found


     NOTICE.* .*: Request \'REGISTER\' from '.*' failed for 
':.*' .* - Failed to authenticate


     NOTICE.* .*: Request \'REGISTER\' from '.*' failed for 
':.*' .* - Error to authenticate


     NOTICE.* .*: Request \'INVITE\' from '.*' failed for 
':.*' .*


John Bittner

Xaccel

[...]

We have this rules:

[INCLUDES] 






# Read common prefixes. If any customizations available -- read them 
from 

# common.local 



before = common.conf 






[Definition] 






_daemon = asterisk 






__pid_re = (?:\s*\[\d+\]) 






iso8601 = \d{4}-\d{2}-\d{2}T\d{2}:\d{2}:\d{2}\.\d+[+-]\d{4} 






# All Asterisk log messages begin like this: 



log_prefix= (?:NOTICE|SECURITY|WARNING)%(__pid_re)s:?(?:\[C-[\da-f]*\])? 
[^:]+:\d*(?:(?: in)? \w+:)? 





prefregex = ^%(__prefix_line)s%(log_prefix)s .+$ 






failregex = ^Registration from '[^']*' failed for '(:\d+)?' - 
(?:Wrong password|Username/auth name mismatch|No matching peer found|Not 
a local domain|Device does not ma
tch ACL|Peer is not supposed to register|ACL error \(permit/deny\)|Not a 
local domain)$ 

^Call from '[^']*' \(:\d+\) to extension '[^']*' 
rejected because extension not found in context 

^(?:Host )? (?:failed (?:to authenticate\b|MD5 
authentication\b)|tried to authenticate with nonexistent user\b) 

^No registration for peer '[^']*' \(from \)$ 



^hacking attempt detected ''$ 




^SecurityEvent="(?:FailedACL|InvalidAccountID|ChallengeResponseFailed|InvalidPassword)"(?:(?:,(?!RemoteAddress=)\w+="[^"]*")*|.*?),RemoteAddress="IPV[46]/(UDP|TCP
|WS)//\d+"(?:,(?!RemoteAddress=)\w+="[^"]*")*$ 



^"Rejecting unknown SIP connection from "$ 



^Request (?:'[^']*' )?from '(?:[^']*|.*?)' failed for 
'(?::\d+)?'\s\(callid: [^\)]*\) - (?:No matching endpoint 
found|Not match Endpoint(?: Contact)? ACL|(?
:Failed|Error) to authenticate)\s*$ 






# FreePBX (todo: make optional in v.0.10): 



# 
^(%(__prefix_line)s|\[\]\s*WARNING%(__pid_re)s:?(?:\[C-[\da-f]*\])? 
)[^:]+: Friendly Scanner from $ 





ignoreregex = 






datepattern = {^LN-BEG} 






# Author: Xavier Devlamynck / Daniel Black

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Re: [asterisk-users] No external audio on SIP => PJSIP both behind same NAT

[Administrator TOOTAI]

Le 08/04/2019 à 20:06, Administrator TOOTAI a écrit :

Hi,

I have following setup: Asterisk 1.4 (IP 10.1.1.250) connect to Asterisk 
13 (IP 10.1.1.251) with PJSIP, this one connected to the provider also 
with PJSIP. Both LAN Asteriks are also connected via IAX.


Everything is working fine except SIP call from 1.4 to external number: 
there is no audio. SIP call to eg demo@Asterisk13 is OK. If I replace 
the SIP link between 1.4 and 13 with the IAX trunk it's OK. Other way is 
OK in full SIP.


I tried few parameters on both Asteriks, no luck. The RTP port range is 
the same on both instance.


If someone had a clue on this, welcome ;) and thanks in advance.


Sorry for noise, I had a typo direct-media=no insteed of direct_media=no

Daniel

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[asterisk-users] No external audio on SIP => PJSIP both behind same NAT

[Administrator TOOTAI]

Hi,

I have following setup: Asterisk 1.4 (IP 10.1.1.250) connect to Asterisk 
13 (IP 10.1.1.251) with PJSIP, this one connected to the provider also 
with PJSIP. Both LAN Asteriks are also connected via IAX.


Everything is working fine except SIP call from 1.4 to external number: 
there is no audio. SIP call to eg demo@Asterisk13 is OK. If I replace 
the SIP link between 1.4 and 13 with the IAX trunk it's OK. Other way is 
OK in full SIP.


I tried few parameters on both Asteriks, no luck. The RTP port range is 
the same on both instance.


If someone had a clue on this, welcome ;) and thanks in advance.

Daniel

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[asterisk-users] Asterisk13 - Dialplan reload does not take modification in account

[Administrator TOOTAI]

Hi all,

I switched an old asterisk 1.8 to a new 13 version, stock version from 
Ubuntu 18.04 server.


I did some modification in dialplan but after a reload they are not 
taken in account :(, even after restarting asterisk.


I checked logs and found lots of

merging incls/swits/igpats from old(localEP) to new(localEP) context, 
registrar = pbx_lua


also lines ending with pbx_config.

What does this mean? Could be the source of my troubles ?

Thanks for any hint

Daniel

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Re: [asterisk-users] pattern matching "+"

[Administrator TOOTAI]

Le 15/03/2019 à 15:18, sean darcy a écrit :

 From my provider I get extensions of:

+1<10digit number>
1<10 digit number>
<10 digit number>

seemingly randomly.

What I'd like to do is

exten=_!1234567890,1,Answer()

which would match anything ending in 1234567890.

But that doesn't work since ! can only be used at the end of a pattern.

I tried

[+\ ][1\ ]1234567890

which didn't work, probably because "\ " means  space, not nothing.

Any suggestions?


exten = _+.,1,Goto(${EXTEN:-10})
exten = _X.,1,Goto(${EXTEN:-10})
...

Daniel

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[asterisk-users] Future of IAX

[Administrator TOOTAI]

Hello,

since ipv6 doesn't really push ipv4 out of our networks, we still use 
IAX (when possible) to not face NAT problems from SIP, Asterisk 16 included.


Question is, what is the future of IAX. Is it a dead protocol ? Will it 
stay as is even only security problems resolved ? Will be (sooner or 
later) EOL ?


Thanks for your informations

Daniel

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Re: [asterisk-users] Asterisk Usage Survey

[Administrator TOOTAI]

Le 11/03/2019 à 10:23, Marcelo Terres a écrit :

Hello Jean-Denis.

I believe the idea is that you answer the survey for each type of 
scenarios you are running.


So one for call centre, another one for ivr, etc...


And what for instance about exact version of asterisk?

We are in the same situation as Jean Denis, running 1.4 to 16 version as 
integrator/service provider/user/...


Difficult to replay the survey for each scenarios ;)



Regards,

Marcelo

On Mon, 11 Mar 2019, 02:10 Jean-Denis Girard, > wrote:


Hi Matt,

I would have loved to participate to the survey, but I feel it does
apply to my situation: as an integrator, I'm installing Asterisk for
call centers, PBX, IVR... so I can not answer the first question of the
survey ;) I also have dfferent versions installed.

This is not a negative comment, I just want to express that the survey
does not seem to apply to me; and many people on the Asterisk lists may
be in a situation similar as mine.


Thanks,
-- 
Jean-Denis Girard


SysNux                   Systèmes   Linux   en   Polynésie  française
https://www.sysnux.pf/   Tél: +689 40.50.10.40 / GSM: +689 87.797.527

Le 08/03/2019 à 05:35, Matthew Fredrickson a écrit :
 > Hey All,
 >
 > For those of you that do not know me, my name is Matthew Fredrickson
 > and I’m the project lead for the Asterisk project. First off, I
wanted
 > to thank all of you that contribute in various ways to the project –
 > whether it be at a developmental level, answering questions on forums
 > and mailing lists, contributing documentation, or just generally
 > advocating for it within your sphere of influence. It takes so many
 > people’s efforts to make the project what it is and to sustain such a
 > large and vibrant user and developer community.
 >
 > We created a general survey inquiring how people utilize Asterisk. It
 > should only take about 10-15 minutes, but would help us understand
 > better how our users are utilizing Asterisk and help us to understand
 > if there are important areas of Asterisk that we underemphasize
from a
 > development perspective. If you don’t mind filling it out, it
would be
 > greatly appreciated.
 >
 > Thanks *so* much again for your time, and best wishes to each of you
 > in your efforts.
 >
 > https://goo.gl/forms/xL1VUHRsf95saly13
 >
 > Matthew Fredrickson
 >

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[asterisk-users] PJSIP IPv6 remote_hosts

[Administrator TOOTAI]

Hi,

I rey to register an Asterisk 16.2.1 pjsip to an ASTERISK 13.25.0 
chan_sip using ipv6 and pjsip_wizard.


I only got it work if in remote_hosts I put the ipv6 address and not the 
hostname like sip.domain.ltd No need to say that an  entry is 
existing for the hostname in DNS.


BTW, how are DNS SRV entry work with ipv6 ?

Thanks for your support

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[asterisk-users] Gigaset C610 IP error with PJSIP

[Administrator TOOTAI]

Hi,

We upgraded an Asterisk 11 server to 16.1.1, going from chan_sip to 
pjsip, on a site using Gigaset phones. They are registring well despite 
the fact that we get a lot of errors like


[Feb 22 18:30:07] ERROR[1556]: pjproject: :  sip_transport. Error 
processing 367 bytes packet from UDP 192.168.1.108:5060 : PJSIP syntax 
error exception when parsing 'To' header on line 4 col 51:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.1.250:5060;rport=5060;branch=z9hG4bKPjf89448cc-aeed-47be-b9a3-4e9b79f064bd
From: 
;tag=9a7c1775-fe8d-4c67-a8a4-c44e5d473742
To: 
;tag=8`6b0664,gd9e,5b76,`9`5,b55d4e562653

Call-ID: fd359d3b-1102-4cd5-929a-46a04a95389d
CSeq: 28722 NOTIFY
Content-Length: 0

It seems that the comma in the  tag is the origin of this behavior.

Is this an Asterisk problem or a Gigaset one ?

Daniel

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Re: [asterisk-users] trouble removing + sign

[Administrator TOOTAI]

Le 14/02/2019 à 00:12, sean darcy a écrit :
I'm using BLACKLIST() to check numbers, which does not like leading + 
signs. I want to test if there is a plus sign, and then remove it.


I tried:

  ;  strip leading plus sign
   same => n, Verbose( callerid 0:1 is ${CALLERID(num):0:1} )
   same => n,ExecIf($["${CALLERID(num):0:1}" = "+"]?Set(CALLERID(num) = 
${CALLERID(num):1})

   same=>n,GotoIf(${BLACKLIST()}?make-em-wait)

but it's stripping the first character + sign or not. The callerid is 
1203XX


     -- Executing [s@hangup-spam:3] Verbose("PJSIP/2667075-000b", " 
callerid 0:1 is 1 ") in new stack

  callerid 0:1 is 1
     -- Executing [s@hangup-spam:4] ExecIf("PJSIP/2667075-000b", 
"0?Set(CALLERID(num) = 203XXX") in new stack
     -- Executing [s@hangup-spam:5] GotoIf("PJSIP/2667075-000b", 
"0?make-em-wait") in new stack


ExecIf correctly finds the comparison false(the "0"), but still executes 
the appiftrue .


What am I missing ?


Try ExecIf($["x${CALLERID(num):0:1}" == "x+"]?Set(CALLERID(num) = 
${CALLERID(num):1})


Or you could use somethjing like

exten = _X.,1,NoOp(Your dialplan)
 same = n,...
exten = _+.,1,Goto(${EXTEN:1},1)

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Re: [asterisk-users] Asterisk on dynamich IP

[Administrator TOOTAI]

Le 01/02/2019 à 09:59, basti a écrit :

Hello,


Hi



my Asterisk is installed on my router. From my ISP I only get an dynamic IP.

In sip.conf I have try:

externhost=host1.mydns.unix-solution.de
externrefresh=300

but after reconnect I cant call from "outside".

asterisk*CLI> sip show registry
Hostdnsmgr Username   Refresh
StateReg.Time
sip.personal-voip.de:5060   N  1785
Registered   Fri, 01 Feb 2019 09:32:28
sip.alice-voip.de:5060  N    1785 Registered
   Fri, 01 Feb 2019 09:32:28
2 SIP registrations.
asterisk*CLI>

I have also try to restart asterisk after reconnect,
Manual it works but not via cron.

Are there any other ways to reconnect on IP change.


I'm surprised that it doesn't work via cron. I suspect the script you 
are running faulty like wrong path or not executed by the right user.


For other purpose that for asterisk I also check IP changes on some 
sites, my script is always returning the correct datas.


Daniel

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Re: [asterisk-users] [Asterisk-video] asterisk playing video call to a local display

[Administrator TOOTAI]

Le 30/01/2019 à 05:17, Jose Tavares a écrit :

Hi guys ..
I have some experience with asterisk and sip since I have been using it 
for over 10 years.
But in the last years I have been just maintaining the installations we 
have without updating myself on the new features of it.


Now I have a requirement that is dialing to a h323 endpoint that answers 
with video and I need to display it in a local display.


I have already tried Ekiga (unstable, difficult to automate since it is 
a graphical app) and linphone that does not support h323. I also found 
simpleopal that I don't know if it can handle that.


My idea is to convert this to a Docker Container later and to run it in 
a Raspberry Pi. I have already been playing with Raspberry Pi and 
containers for some other purposes with good results..


So, I would like to know if asterisk offers a way to automatically 
listen to a system event, dial to a h323 endpoint that will answer with 
video and if it has drivers to display locally the video or to pipe to a 
omxplayer, for example.


chan_ooh323 does not do the job ?

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Re: [asterisk-users] [OT] Are anonymous international calls allowed ?

[Administrator TOOTAI]

Le 17/01/2019 à 10:38, Olivier a écrit :

Hello,


Hi



These questions crossed my mind this morning :
In general, are anonymous international calls allowed (ie calling from 
one country to a number in an other country while hiding your own caller 
id) ?

Are there special rules in Europe for this ?


Please define an anonymous call ? Is it  as CID ?

Rules are provider specific: some let's you change your CID other not 
(remember Freephonie in France, they didn't allow changing CID). I know 
other providers -especially in the UK- who are changing the CID if the 
caller has an CID which is not a UK one and the call is going abroad.


In Europe, most countries are asking to have an address in the country. 
France is asking but not checking. Germany and Italy are checking the 
postal address you gave.


Anyway, Anonymous can be a name, people not speaking English will not 
know what it means. And for the number, I think putting something like 
0123456 could be enough, so far I tested, nobody is asking you to put 
the number is international format. Even, some Telco replace your DID 
with the one you have by them in national format doesn't matter if 
you're calling abroad !


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[asterisk-users] Out of queue - no pickup after 0ms

[Administrator TOOTAI]

Hello,

I have an external agent which register dynamically in a queue and I 
setup his PJSIP account on a identity of a local phone which is 
configured to redirect all calls to the agent. Agent can have only on 
call at a time.


When agent takes a call he is paused from the queue to avoid further 
dial from the queue has his state is always "Not in Use". What happend 
is that after having taking a call, the agent is unpaused and should get 
next call in the queue. But what I see is


app_queue.c: Called PJSIP/exten115
app_queue.c: -- PJSIP/exten115-0620 connected line has changed. 
Saving it until answer for PJSIP/TOOTAiAudio-0612

app_queue.c: -- Nobody picked up in 0 ms
 [many times]

What does this mean ? TOOTAiAudio-0612 being the caller channel in 
the queue, the callee channel increase on each try. The callee is well 
known to be idle.


app_queue.c: Called PJSIP/exten115
app_queue.c: -- PJSIP/exten115-0621 connected line has changed. 
Saving it until answer for PJSIP/TOOTAiAudio-0612

app_queue.c: -- Nobody picked up in 0 ms
app_queue.c: Called PJSIP/exten115
app_queue.c: -- PJSIP/exten115-0622 connected line has changed. 
Saving it until answer for PJSIP/TOOTAiAudio-0612

app_queue.c: -- Nobody picked up in 0 ms
app_queue.c: Called PJSIP/exten115
app_queue.c: -- PJSIP/exten115-0623 connected line has changed. 
Saving it until answer for PJSIP/TOOTAiAudio-0612

app_queue.c: -- Nobody picked up in 0 ms

delay between tries being around 5 seconds.

Other question, is it possible to tell to a queue that a member is in use ?

Asterisk version is 16.1.1

Thanks for any hint

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Re: [asterisk-users] Detecting a fax

[Administrator TOOTAI]

Le 11/01/2019 à 10:23, Neil Youngman a écrit :

On 11/01/2019 09:19, Administrator TOOTAI wrote:

Le 11/01/2019 à 10:12, Neil Youngman a écrit :
A while back, I posted about detecting when a call was picked up by 
a fax machine. It was suggested that having a "fax" extension and 
"faxdetect=yes" would cause it to jump to the "fax" extension. This 
was not something I could get to work.


I have now created a very simple example. In sip.conf I have 
"faxdetect = yes". My example extension is:


[test]
;
; Voice test extension
;
exten => voicetest,1,NoOp()
 same => n,LOG(Notice,${CHANNEL}: Extension voiceout starting)
 same => n,LOG(Notice,${CHANNEL}: Starting Answer Machine Detection)
 same => n,AMD()
 same => n,LOG(Notice,${CHANNEL}: Answer Machine Detection 
${AMDSTATUS}/${AMDCAUSE})

 same => n,Playback(/var/lib/asterisk/sounds/en/demo-congrats)
 same => n,LOG(Notice,${CHANNEL}: Voice out extension complete)
 same => n(hangup),Hangup()


;
; Fax detected extension
;
exten => fax,1,NoOp()
 same => n,LOG(Notice,${CHANNEL}: Extension fax starting)
 same => n,LOG(Notice,${CHANNEL}: Fax Machine Detected)
 same => n,Playback(/var/lib/asterisk/sounds/en/silence/2)
 same => n,LOG(Notice,${CHANNEL}: Fax extension complete)
 same => n(hangup),Hangup()

and the logs show that calling a fax using the voiceout extension in 
context test does not result in the fax extension being triggered.


[Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Extension voiceout starting
[Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Starting Answer Machine Detection
[Jan 11 08:55:13] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Answer Machine Detection 
MACHINE/LONGGREETING-1500-1500
[Jan 11 08:55:44] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Voice out extension complete


Just for completeness this is how the call is originated, with a 
different phone number:


Action: Originate
ActionId: 1234567W001-125
Context: test
Exten: voicetest
Priority: 1
Channel: SIP/+441632660987@31.13.156.183:5060
Timeout: 6
Async: True

Can anyone offer any insight into why this isn't working?

Neil Youngman




You didn't ANSWER() the call


It's an outgoing call. I wouldn't expect to answer an outgoing call?


I don't understand your goal. You want to send or receive fax?

On our side, dialplan very simplified on incoming calls (receive fax):

. in CLI> fax show settings ; output FAX For Asterisk Settings: blabla
. Set(FAXOPT(faxdetect)=yes); or faxdetect=yes in sip.conf
. Answer()
; if a fax is detected, go to fax extension
. exten => fax,n,ReceiveFAX(${FAXPATHTOFILE}/${FAXFILE},fs)

and done.
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Re: [asterisk-users] Detecting a fax

[Administrator TOOTAI]

Le 11/01/2019 à 10:12, Neil Youngman a écrit :
A while back, I posted about detecting when a call was picked up by a 
fax machine. It was suggested that having a "fax" extension and 
"faxdetect=yes" would cause it to jump to the "fax" extension. This was 
not something I could get to work.


I have now created a very simple example. In sip.conf I have "faxdetect 
= yes". My example extension is:


[test]
;
; Voice test extension
;
exten => voicetest,1,NoOp()
     same => n,LOG(Notice,${CHANNEL}: Extension voiceout starting)
     same => n,LOG(Notice,${CHANNEL}: Starting Answer Machine Detection)
     same => n,AMD()
     same => n,LOG(Notice,${CHANNEL}: Answer Machine Detection 
${AMDSTATUS}/${AMDCAUSE})

     same => n,Playback(/var/lib/asterisk/sounds/en/demo-congrats)
     same => n,LOG(Notice,${CHANNEL}: Voice out extension complete)
     same => n(hangup),Hangup()


;
; Fax detected extension
;
exten => fax,1,NoOp()
     same => n,LOG(Notice,${CHANNEL}: Extension fax starting)
     same => n,LOG(Notice,${CHANNEL}: Fax Machine Detected)
     same => n,Playback(/var/lib/asterisk/sounds/en/silence/2)
     same => n,LOG(Notice,${CHANNEL}: Fax extension complete)
     same => n(hangup),Hangup()

and the logs show that calling a fax using the voiceout extension in 
context test does not result in the fax extension being triggered.


[Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Extension voiceout starting
[Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Starting Answer Machine Detection
[Jan 11 08:55:13] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Answer Machine Detection 
MACHINE/LONGGREETING-1500-1500
[Jan 11 08:55:44] NOTICE[18073][C-0115] Ext. voicetest: 
SIP/31.13.156.183:5060-00f4: Voice out extension complete


Just for completeness this is how the call is originated, with a 
different phone number:


Action: Originate
ActionId: 1234567W001-125
Context: test
Exten: voicetest
Priority: 1
Channel: SIP/+441632660987@31.13.156.183:5060
Timeout: 6
Async: True

Can anyone offer any insight into why this isn't working?

Neil Youngman




You didn't ANSWER() the call

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Re: [asterisk-users] Hint and state

[Administrator TOOTAI]

Le 10/01/2019 à 16:18, Social Boh a écrit :

Hello,

maybe:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_DEVICE_STATE 


Thanks Social, exactly what I wanted.




---
I'm SoCIaL, MayBe

El 10/01/2019 a las 09:13, Administrator TOOTAI escribió:

Hi,

on an Asterisk 16 with PJSIP I want to know the state of a device 
(idle, busy, unavailable, ...) in the dialplan. I tried with 
ChanIsAvail() but this one doesn't return the real state (eg a device 
calling an extension which is running ChanIsAvail() is marked as idle!)


When I use in a console "core show hints" or "core show hint 
" I get the right information. How to get the same 
information in a dialplan ?


Thanks for any hint (ha ha ;))

Daniel





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[asterisk-users] Hint and state

[Administrator TOOTAI]

Hi,

on an Asterisk 16 with PJSIP I want to know the state of a device (idle, 
busy, unavailable, ...) in the dialplan. I tried with ChanIsAvail() but 
this one doesn't return the real state (eg a device calling an extension 
which is running ChanIsAvail() is marked as idle!)


When I use in a console "core show hints" or "core show hint 
" I get the right information. How to get the same 
information in a dialplan ?


Thanks for any hint (ha ha ;))

Daniel

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Re: [asterisk-users] Pjsip and Call limit

[Administrator TOOTAI]

Le 27/12/2018 à 20:42, Social Boh a écrit :

Hello,

you have to use GROUP and GROUP_COUNT functions.


Well, could be done for extensions but for queue ? Does it mean 
ringinuse is useless ?



[...]
El 27/12/2018 a las 14:14, Administrator TOOTAI escribió:

Hello,

I'm used to set call-limit in sip.conf Now I switched one customer 
Asterisk to 16 version and can't get the behavior back, as well for 
extensions as for queues.


I set ringinuse=no for queues and have max_audio_streams = 1 
max_video_streams = 0. I wanted to add max_calls = 1 but this 
parameter is not accepted.


Thanks for any hint


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[asterisk-users] Pjsip and Call limit

[Administrator TOOTAI]

Hello,

I'm used to set call-limit in sip.conf Now I switched one customer 
Asterisk to 16 version and can't get the behavior back, as well for 
extensions as for queues.


I set ringinuse=no for queues and have max_audio_streams = 1 
max_video_streams = 0. I wanted to add max_calls = 1 but this parameter 
is not accepted.


Thanks for any hint

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Re: [asterisk-users] how to use a database

[Administrator TOOTAI]

Le 07/12/2018 à 15:56, hw a écrit :

On 12/07/2018 03:36 PM, Administrator TOOTAI wrote:

Le 07/12/2018 à 14:32, hw a écrit :

[...]


Queues seem to be the only way to have several phones ring at once, 
or are there other ways?


Dial(SIP/Phone1&SIP/Phone2&...&SIP/Phonex,,)



Good to know, thanks!


What are the entries needed in the queue_members table when using odbc? 
Alembic made the primary key so that each queue can only have one entry 
(What is an interface here?), and there's probably a reason for that. 
How do you enter several members for a queue?  Asterisk seems to either 
rather crash than to create a queue, or to do nothing.


Why you don't just add members dynamically in a queu using 
AddQueueMember/RemoveQueueMember or even with pause/unpause members ?


BTW the above dial string has nothing to do with queue, it just a cmd 
that rings all phones at once.


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Re: [asterisk-users] how to use a database

[Administrator TOOTAI]

Le 07/12/2018 à 14:32, hw a écrit :

[...]


Queues seem to be the only way to have several phones ring at once, or 
are there other ways?


Dial(SIP/Phone1&SIP/Phone2&...&SIP/Phonex,,)

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Re: [asterisk-users] PJSIP add header on forwarded call

[Administrator TOOTAI]

Le 27/11/2018 à 13:18, Joshua C. Colp a écrit :

On Tue, Nov 27, 2018, at 8:13 AM, Administrator TOOTAI wrote:

Le 27/11/2018 à 12:13, Joshua C. Colp a écrit :

On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...]


[TOOTAiAudio]
;
; Call our gateway

exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1})
    same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T)
    same = n,Return

exten = h,1,NoOp()
    same = n,NoOp(Hangup Cause: ${HANGUPCAUSE})
    same = n,NoOp(Dial status : ${DIALSTATUS})
    same = n,NoOp(X-TOOTAiAudio=${PJSIP_HEADER(read,X-TOOTAiAudio-CALLED)})
    same = n,Return

[...]


Why can't be PJSIP extra headers setted in this case ?


As documented on the wiki[1] the PJSIP_HEADER dialplan function has to be 
executed on the PJSIP channel itself, not the calling channel. You need to use 
a pre-dial handler and invoke it there.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_HEADER



Thanks Joshua, that worked. As you see above I want to have the value of
headers when call is ended. Problem is that on h extension the channel
already gone.

Is there a solution to archieve this ?


Is there a reason you can't use a normal dialplan variable instead?


That's what I do at this time. I thought I could bypass this by 
retriving the output of headers


 Otherwise I don't believe PJSIP_HEADER will retrieve such information 
regardless, it's for querying headers on an incoming INVITE.




Ok, thanks for your help.

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Re: [asterisk-users] PJSIP add header on forwarded call

[Administrator TOOTAI]

Le 27/11/2018 à 12:13, Joshua C. Colp a écrit :

On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...]


[TOOTAiAudio]
;
; Call our gateway

exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1})
   same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T)
   same = n,Return

exten = h,1,NoOp()
   same = n,NoOp(Hangup Cause: ${HANGUPCAUSE})
   same = n,NoOp(Dial status : ${DIALSTATUS})
   same = n,NoOp(X-TOOTAiAudio=${PJSIP_HEADER(read,X-TOOTAiAudio-CALLED)})
   same = n,Return

[...]


Why can't be PJSIP extra headers setted in this case ?


As documented on the wiki[1] the PJSIP_HEADER dialplan function has to be 
executed on the PJSIP channel itself, not the calling channel. You need to use 
a pre-dial handler and invoke it there.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_HEADER



Thanks Joshua, that worked. As you see above I want to have the value of 
headers when call is ended. Problem is that on h extension the channel 
already gone.


Is there a solution to archieve this ?

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[asterisk-users] PJSIP add header on forwarded call

[Administrator TOOTAI]

Hi list,

to manage an external queue agent the only solution I found is to 
connect a local account and redirect calls to this account using forward 
features from the phone (SNOM). The problem I face is that before 
calling the agent I would like to set extra header. Dialplan to call 
external agent is this one with (Gosub):


[TOOTAiAudio]
;
; Call our gateway

exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1})
 same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T)
 same = n,Return

exten = h,1,NoOp()
 same = n,NoOp(Hangup Cause: ${HANGUPCAUSE})
 same = n,NoOp(Dial status : ${DIALSTATUS})
 same = n,NoOp(X-TOOTAiAudio=${PJSIP_HEADER(read,X-TOOTAiAudio-CALLED)})
 same = n,Return

When a local phone call extension 115 (the one where calls to external 
agent are forwarded), everything is working well. But if I call the 
account from a queue I get



[Nov 27 09:54:08] ERROR[12758][C-005f]: res_pjsip_header_funcs.c:513 
func_write_header: This function requires a PJSIP channel


Output of queue is

deblix9*CLI> queue show q301
q301 has 0 calls (max unlimited) in 'ringall' strategy (6s holdtime, 47s 
talktime), W:0, C:25, A:3, SL:100.0%, SL2:100.0% within 60s

   Members:
  PJSIP/PPermis115 (ringinuse disabled) (dynamic) (Not in use) has 
taken 8 calls (last was 706 secs ago)

   No Callers

where PPermis115 is the local account on a phone who forward calls to 
extension 115.


Why can't be PJSIP extra headers setted in this case ?

Thanks for any hint

Reagrds

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Re: [asterisk-users] Queue member not local - PJSIP - Asterisk 16

[Administrator TOOTAI]

No one on this ?

Le 22/11/2018 à 17:59, Administrator TOOTAI a écrit :

Hi all,

I want to set dynamic queue with non local members. I create an 
extension 115 in [localEP] context which is doing the job, eg calls to 
this extension are forwarded to the non local endpoint (which is an IP 
phone connected to an external Asterisk 13 version). Phones are SNOM.


Queue looks like this (all members defines the same one, test purpose):

deblix9*CLI> queue show q301
q301 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s 
talktime), W:0, C:0, A:18, SL:0.0%, SL2:94.4% within 60s

    Members:
   PJSIP/TOOTAI115@TOOTAiAudio (ringinuse disabled) (dynamic) 
(Invalid) has taken no calls yet
   PJSIP/PPermis115 (ringinuse disabled) (dynamic) (Not in use) has 
taken no calls yet
   Local/115@localEP/n (ringinuse disabled) (dynamic) (Invalid) has 
taken no calls yet

    No Callers

where Local/115 is the working extension I spoke above. The 
PJSIP/TOOTAI115 being the external member. If I display DEVICE_STATE in 
dialplan, I get the INVALID status as shown above.


I also tried to setup an PPermis115 peer in a phone and modify features 
to forward all calls. This doesn't work either getting below about when 
calling the queue:


PJSIP/PPermis115-00d3 connected line has changed. Saving it until 
answer for PJSIP/PPermis102-00d1

     -- Forwarding PJSIP/PPermis102-00d1 to '125' prevented.
[continuously]

Is there a way to force the state of a member or to tell to a queue to 
call a member anyway even if the state is invalid? Other solution?


Thanks for any hint

Regards



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[asterisk-users] Queue member not local - PJSIP - Asterisk 16

[Administrator TOOTAI]

Hi all,

I want to set dynamic queue with non local members. I create an 
extension 115 in [localEP] context which is doing the job, eg calls to 
this extension are forwarded to the non local endpoint (which is an IP 
phone connected to an external Asterisk 13 version). Phones are SNOM.


Queue looks like this (all members defines the same one, test purpose):

deblix9*CLI> queue show q301
q301 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s 
talktime), W:0, C:0, A:18, SL:0.0%, SL2:94.4% within 60s

   Members:
  PJSIP/TOOTAI115@TOOTAiAudio (ringinuse disabled) (dynamic) 
(Invalid) has taken no calls yet
  PJSIP/PPermis115 (ringinuse disabled) (dynamic) (Not in use) has 
taken no calls yet
  Local/115@localEP/n (ringinuse disabled) (dynamic) (Invalid) has 
taken no calls yet

   No Callers

where Local/115 is the working extension I spoke above. The 
PJSIP/TOOTAI115 being the external member. If I display DEVICE_STATE in 
dialplan, I get the INVALID status as shown above.


I also tried to setup an PPermis115 peer in a phone and modify features 
to forward all calls. This doesn't work either getting below about when 
calling the queue:


PJSIP/PPermis115-00d3 connected line has changed. Saving it until 
answer for PJSIP/PPermis102-00d1

    -- Forwarding PJSIP/PPermis102-00d1 to '125' prevented.
[continuously]

Is there a way to force the state of a member or to tell to a queue to 
call a member anyway even if the state is invalid? Other solution?


Thanks for any hint

Regards

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Re: [asterisk-users] Asterisk 16 PJSIP and set_var

[Administrator TOOTAI]

Le 20/11/2018 à 19:50, Administrator TOOTAI a écrit :

Hi,

I'm on the way to upgrade a dialplan from 1.8 to 16.0.1 and face a 
problem with user variable defined in sip.conf using setvar. It work 
like a charm -even on asterisk 13 version- but can't get it work in 16. 
The variables are defined in pjsip with set_var and a pjsip show 
endpoint  does show them, like


myMailMonitor  :
myOnNOANSWER   :  main
myPrivateEnv   :  PPermis
myPrivateVM    :  yes
mySubscriptions    :  10

In dialplan -the same that 1.8- I have

exten => _X.,n,Set(__DIAL_OPTIONS=rT)
exten => _X.,n,Set(__PrivateEnv=${myPrivateEnv})
exten => _X.,n,Set(__PrivateVM=${myPrivateVM})
exten => _X.,n,Set(__OnNOANSWER=${myOnNOANSWER})
exten => _X.,n,Set(__MailMonitor=${myMailMonitor})

but variables are empty. Is there a new way to recover those values ?


Got it, problem is in set_var conf, it must not have any space between 
variable name, equal sign and value


Working: set_var = myPrivateEnv=
Not working: set_var = myPrivateEnv = 

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[asterisk-users] Asterisk 16 PJSIP and set_var

[Administrator TOOTAI]

Hi,

I'm on the way to upgrade a dialplan from 1.8 to 16.0.1 and face a 
problem with user variable defined in sip.conf using setvar. It work 
like a charm -even on asterisk 13 version- but can't get it work in 16. 
The variables are defined in pjsip with set_var and a pjsip show 
endpoint  does show them, like


myMailMonitor  :
myOnNOANSWER   :  main
myPrivateEnv   :  PPermis
myPrivateVM    :  yes
mySubscriptions    :  10

In dialplan -the same that 1.8- I have

exten => _X.,n,Set(__DIAL_OPTIONS=rT)
exten => _X.,n,Set(__PrivateEnv=${myPrivateEnv})
exten => _X.,n,Set(__PrivateVM=${myPrivateVM})
exten => _X.,n,Set(__OnNOANSWER=${myOnNOANSWER})
exten => _X.,n,Set(__MailMonitor=${myMailMonitor})

but variables are empty. Is there a new way to recover those values ?

Thanks for any hint

Regards

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[asterisk-users] Forward call to another device or etxtension

[Administrator TOOTAI]

Hi list,

I have a queue in which I add a member located outside the company and 
connected to an outside asterisk. Let's say peername is ABCD123. In the 
queue I gave SIP/ABCD123 as interface which is not existing on the local 
asterisk.


Is there a way to connect a member from a queue which is not known from 
asterisk (eg when queue call SIP/ABCD123 a forwarding feature send it to 
an extension or other device) ?


In my case, an extension 123@oneContext call the outside device so I 
found a solution by using dynamic agent like 
AddQueueMember(myqueue,Local/123@oneContext). But for my knowledge still 
would know if my request is possible and has some sense :) Would the 
atsdb have some feature I could use ?


Thanks for any hint.

Daniel

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Re: [asterisk-users] SHELL() function Asterisk 13 - can only accept one paramter in string?

[Administrator TOOTAI]

Le 27/07/2018 à 09:36, Stefan Viljoen a écrit :

Hi all

This is a followup on my post "Asterisk 13 - system() dialplan app cannot call bash 
scripts" from yesterday

I've given up trying to use system() to call BASH scripts with parameters from 
Asterisk 13.

Turned out under Asterisk 13.22.0 System() DOES work, but only if you do NOT 
attempt to pass any parameters to the called script.


[...]

*CLI> core show version
Asterisk 13.22.0 built by root @ pabx on a x86_64 running Linux on 
2018-07-14 13:36:49 UTC


This works for us

same = n,system(/bin/echo "To: ${CALLED_CHANNEL}   From: ${ORI_CALL}" | 
/usr/bin/mail -s "TOOTAiAudio - Congestion Gateway ${CONGESTION_GW} 
used" ${AUDIO_ADMIN})


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[asterisk-users] No register between Asterisk 15 and 13 running pjsip

[Administrator TOOTAI]

Hello,

we have 4 asteriks, 2 in office on one server (wazo and mobydick), and 2 
in DC (self compiled) each on his own server. All of them are VMs under 
Debian Stretch. We used OpenVPN to connect the machines together in TAP 
mode, everything was running well.


Setup is following: the 2 asterisk in office on the same server are 
Asterisk 15 (wazo) and Asterisk 11certified (mobydick). Each of them is 
connected to the 2 others Asterisk in DC, both being Asterisk 13, all 
using chan_sip except one in DC wich is pjsip. They are also 2 IP phones 
in the office which are connected to all servers. As stated above, in 
tap mode everything is running well.


Now we changed our VPNs to use tun. The setup was tested appart of 
asterisks, all connections are OK, all machines can speak to each others 
including Windows one.


Now the problem: all VOIP devices are connecting as before except the 
Asterisk 15 from Office who can't register to the Asterisk 13 in DC 
running pjsip. No problem with the Asterisk11 certified against the same 
pjsip, as well as no problem to the other Asterisk 13 in DC running 
chan_sip.


What we get:

<--- Received SIP request (394 bytes) from UDP:10.99.0.52:5060 --->
REGISTER sip:zone-s SIP/2.0
Via: SIP/2.0/UDP 192.168.12.250:5060;branch=z9hG4bK4d838884
Max-Forwards: 70
From: ;tag=as17aa56c4
To: 
Call-ID: 2efc5e2320a31ff1107505663a02397d@127.0.1.1
CSeq: 102 REGISTER
Supported: replaces, timer
User-Agent: Office PBX
Expires: 3600
Contact: 
Content-Length: 0


[2018-07-05 18:49:08] NOTICE[21317]: acl.c:750 ast_apply_acl: SIP ACL: 
Rejecting '10.99.0.52' due to a failure to pass ACL '(BASELINE)'
[2018-07-05 18:49:08] NOTICE[21317]: res_pjsip/pjsip_distributor.c:649 
log_failed_request: Request 'REGISTER' from '' 
failed for '10.99.0.52:5060' (call$

d: 2efc5e2320a31ff1107505663a02397d@127.0.1.1) - Not match Endpoint ACL
<--- Transmitting SIP response (322 bytes) to UDP:10.99.0.52:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 
192.168.12.250:5060;rport=5060;received=10.99.0.52;branch=z9hG4bK4d838884

Call-ID: 2efc5e2320a31ff1107505663a02397d@127.0.1.1
From: ;tag=as17aa56c4
To: ;tag=z9hG4bK4d838884
CSeq: 102 REGISTER
Server: TOOTAiAudio
Content-Length:  0

where 10.99.0.52 is the IP of the office tun VPN and 192.168.12.250 is 
the Asterisk 15 IP. zone-s is the hostname of the Asterisk pjsip server. 
The 10.99.0.52 is not in ACL (we tried by including it but no luck). 
zwr-IPBX is the username/auth_user. Remember, both VOIP phones as well 
as the Asterisk 11 server connect without problem. The Asterisk 11 an 
Asterisk 13 configuration is the same in pjsip.conf appart of 
username/auth_name.


If someone had any clue on this.

Regards

Daniel

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[asterisk-users] Possibility to access PJSIP variables from dialplan

[Administrator TOOTAI]

Hi all,

is it possible to access PJSIP configuration variables from the dialplan 
? Exemple: I want to get the username of a type = auth context.


Thanks for any hint

Daniel

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Re: [asterisk-users] PJSIP error No auth credentials for realm(s) 'asterisk' in challenge

[Administrator TOOTAI]

Le 16/04/2018 à 16:52, Joshua Colp a écrit :

On Mon, Apr 16, 2018, at 11:47 AM, Administrator TOOTAI wrote:

Hi all,

we are trying to move our servers from chan_sip to chan_pjsip. At this
time no problems with phones, they all register fine and can place
calls. But for a trunk we face problem and can't place calls despite the
fact that registration is OK. What we get is:

[2018-04-16 16:08:33] WARNING[18665]:
res_pjsip_outbound_authenticator_digest.c:178
digest_create_request_with_auth_from_old: Endpoint: 'sip.xxx.tld':
Unable to create request with auth. No auth credentials for realm(s)
'asterisk' in challenge.


The remote side challenged for authentication but your endpoint has no 
"outbound_auth" configured, so chan_pjsip has no idea of how to authenticate.



Thanks Joshua, that did it. We already tested a sort of by inserting 
line = yes and endpoint = sip.xxx.tld in registration stanza but this 
didn't work


Again, many thanks.

Daniel

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[asterisk-users] PJSIP error No auth credentials for realm(s) 'asterisk' in challenge

[Administrator TOOTAI]

Hi all,

we are trying to move our servers from chan_sip to chan_pjsip. At this 
time no problems with phones, they all register fine and can place 
calls. But for a trunk we face problem and can't place calls despite the 
fact that registration is OK. What we get is:


[2018-04-16 16:08:33] WARNING[18665]: 
res_pjsip_outbound_authenticator_digest.c:178 
digest_create_request_with_auth_from_old: Endpoint: 'sip.xxx.tld': 
Unable to create request with auth. No auth credentials for realm(s) 
'asterisk' in challenge.


Our setup:

[sip.xxx.tld]
type = registration
retry_interval = 20
max_retries = 0
contact_user = 
expiration = 3600
transport = transport-udp
outbound_auth = sip.xxx.tld
client_uri = sip:@sip.xxx.tld
server_uri = sip:sip.xxx.tld

[sip.xxx.tld]
type = auth
password = 
username = 

[sip.xxx.tld]
type = identify
endpoint = sip.xxx.tld
match = 

[sip.xxx.tld]
type = endpoint
context = from-xxx.tld
aors = sip.xxx.tld
deny = 0.0.0.0/0.0.0.0
permit = /32
permit = /32
dtmf_mode = rfc4733
disallow = all
allow = alaw,ulaw,g729

[sip.xxx.tld]
type = aor
contact = sip::5060

Registry:

zone-s*CLI> pjsip list registrations

  
  

==

 sip.xxx.tld/sip:sip.xxx.tld sip.xxx.tld Registered

Objects found: 1

PJSIP is listening on port 12345, chan_sip on port 5060. The peer end is 
a Kamailio 3.3.4 if it matter.


What could be the problem? Does anyone have a PJSIP asterisk registered 
against Kamailio?


Thanks for any hints

Daniel

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Re: [asterisk-users] Digium IP Phones UNREACHABLE after registration

[Administrator TOOTAI]

Hi Herrmann

Le 12/04/2018 à 17:22, Hermann Wecke a écrit :

I'm trying to solve a mystery for the last couple of days.

I have a mix of D70, D50 and D40 behind NAT. Server is in a
colocation, not a VPS.

For several years, everything was working fine, no issues. A few days
ago I started having problems at one particular site. NO CHANGES have
been made to this office network - same router, switch and internet
provider. No new equipment added or configuration changed (I only
upgraded the firmware and asterisk trying to solve the problem).

A few seconds after registration, the Digium phones will become
UNREACHABLE. Right after that, the entire VoIP network (where the
Digiums are located) will be also dropped - all other devices
(non-Digium) connected will be kicked from the asterisk box. There are
ObiHai, Yealink and Linksys at this location - all will be kicked.

The server remains operational and all other users/peers (not running
Digium phones) are up and running.


Don quite understand: above you say that all others phones are kicked 
too and here you say they are up and running ... Also, UNREACHABLE and 
kick are not the same for me.




Sip debug and tcpdump didn't show any relevant information to solve
the puzzle.


What is relevant: do you see SIP traffic coming from those 
phones/location *AFTER* they are UNREACHABLE/kicked?


Is a fail2ban involved ?

[...]

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Re: [asterisk-users] Client Asterisks can't connect when main Asterisk reboot

[Administrator TOOTAI]

Le 26/03/2018 à 11:08, Antony Stone a écrit :

On Monday 26 March 2018 at 10:58:01, Administrator TOOTAI wrote:


Hi all,

we are running some Asteriks (version 13 on Debian Stretch) as VM/kvm in
datacenter and have trunks with other Asterisk (v1.8 11 or 13) instances
behind FW. Problem we face is that when we reboot the DC Asterisks, the
trunks (SIP or IAX) become alive from DC Asterisks to clients ones but
UNAVAILABLE the other way.

In clients logs we see

Registration for 'XXX@ourDCAsteriks' timed out, trying again (Attempt
#5550)

or UNREACHABLE in case of no registration or IAX

Sometimes a restart from client Asterisks is enough (SIP or IAX reload
has no effect) to recover the full trunk, but in most cases we have to
reboot the client server :(

If we use a VPN (OpenVPN) between Asterisks, no problem.

What can be the problem ?


Are you doing NAT between the two servers?  Your comment about using a VPN
makes me think you probably are.


Yes, NAT is involved.



In that case I suggest the problem is with stale connection tracking table
entries on the machine which is doing the NAT.


OK, effectively the clients having problems are behind ISP boxes which 
means poor NAT level :( Others behind linux boxes (except one behind 
Sophos UTM) doesn't have this problem.


Thanks for your enlightenment.

Daniel

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[asterisk-users] Client Asterisks can't connect when main Asterisk reboot

[Administrator TOOTAI]

Hi all,

we are running some Asteriks (version 13 on Debian Stretch) as VM/kvm in 
datacenter and have trunks with other Asterisk (v1.8 11 or 13) instances 
behind FW. Problem we face is that when we reboot the DC Asterisks, the 
trunks (SIP or IAX) become alive from DC Asterisks to clients ones but 
UNAVAILABLE the other way.


In clients logs we see

Registration for 'XXX@ourDCAsteriks' timed out, trying again (Attempt #5550)

or UNREACHABLE in case of no registration or IAX

Sometimes a restart from client Asterisks is enough (SIP or IAX reload 
has no effect) to recover the full trunk, but in most cases we have to 
reboot the client server :(


If we use a VPN (OpenVPN) between Asterisks, no problem.

What can be the problem ?

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Re: [asterisk-users] how do i enable call features??

[Administrator TOOTAI]

Le 25/01/2018 à 10:37, Atux Atux a écrit :
Being honest, i did not manage to make it work. Now whoever calls the 
system extensions, does not know if they are on another phone call or 
away from the office.


For chan_sip you can do like this before ringing an extension. Status is 
returned in EXTENSTATUS so you can play any announce you want.


Also, il you have 2 calls and the callee doesn't hear a tone about a 
second call, you should take a look in your device configuration.


ARG1=Extension to call
ARG2=should we as well test the extension with IAX tech

[macro-isExtenAvailable] 






exten => s,1,Set(__myEXTEN=${ARG1}) 



exten => s,n,Set(__IAXaction=${ARG2}) 



exten => s,n,Set(__CALLTECH=SIP) 



exten => s,n(checkStatus),ChanIsAvail(${CALLTECH}/${myEXTEN},s) 
   ;channel to test 

exten => s,n,NoOp(Status is  
) 

exten => s,n,Set(__EXTENSTATUS=inuse) 



exten => s,n,GotoIf($["${AVAILSTATUS}" = "2"]?End) 
   ;2 and more are in use status 

exten => s,n,Set(__EXTENSTATUS=busy) 



exten => s,n,GotoIf($["${AVAILSTATUS}" = "3"]?End) 



exten => s,n,Set(__EXTENSTATUS=invalid) 



exten => s,n,GotoIf($["${AVAILSTATUS}" = "4"]?checkIAX) 



exten => s,n,Set(__EXTENSTATUS=unavailable) 



exten => s,n,GotoIf($["${AVAILSTATUS}" = "5"]?checkIAX) 



exten => s,n,Set(__EXTENSTATUS=unavailable) 



exten => s,n,GotoIf($["${AVAILSTATUS}" = "20"]?checkIAX) 



exten => s,n,Set(__EXTENSTATUS=inuse) 



exten => s,n,GotoIf($[${AVAILSTATUS} > 5]?End) 



exten => s,n,Set(__EXTENSTATUS=idle) 



exten => s,n(End),MacroExit 






exten => s,n(checkIAX),GotoIf($["${IAXaction}" = "NOIAX"]?End) 



exten => s,n,GotoIf($["${IAXaction}" = "IAXdone"]?End) 



exten => s,n,Set(__CALLTECH=IAX2) 



exten => s,n,Set(__IAXaction=IAXdone) 
   ;reset ARG2 


exten => s,n,Goto(checkStatus)


Daniel



On Tue, Jan 16, 2018 at 12:30 PM, Atux Atux > wrote:


at the moment i have in each extension in sip.conf the call-limit=2.
Everytime someone calls that extension and that extension is busy,
there is not any notification:
- to the extension that there is a second call
-to the calling party that this extension is on call. So the calling
can either wait or hang up.


How can i make that happen, please?

On Thu, Jan 11, 2018 at 9:58 AM, Atux Atux mailto:atuxn...@gmail.com>> wrote:

No idea on how to write it in my system.

On Thu, Jan 11, 2018 at 12:17 AM, John Kiniston
mailto:johnkinis...@gmail.com>> wrote:

There's some example code in the Dial-Users context of the
basic-pbx samples that might be of use in implementing it.

They are checking a DEVICE_STATE to see if a phone is BUSY,
You could change it to be a database call or implement
custom device states and check those.

wrapping your test case in an ExecIF statement that uses the
DB_EXISTS function to see if the database field you are
checking is valid so you don't get errors about non existent
entries.


https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB_EXISTS



https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_DB


On Wed, Jan 10, 2018 at 11:19 AM, Atux Atux
mailto:atuxn...@gmail.com>> wrote:

That is the general idea. But how do i make it work? is
there somewhere ready?


On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston
mailto:johnkinis...@gmail.com>>
wrote:

Define your *72 and *73 extensions in your internal
context, Have them set a value in the ASTDB that you
then check when dialing your handsets.

The same can be done for call forwarding, store a
number in the ASTDB and check if it's present, if it
is forward the call to that number.

On Wed, Jan 10, 2018 at 12:18 AM, Atux Atux
mailto:atuxn...@gmail.com>> wrote:

Hi. i am running asterisk 11 and i would like to
have features access codes in my system such as
call waiting(all types) (enable/disable), call
forward (enable/disable) and DND. my dialplan is
pretty simple and it is the following

|[DefaultPlan] exten =>
_XX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)
exten => _XX,1,Busy() exten =>
_4XX,2,Answer() exten =>
_4XX,3,VoiceMail(${EXTEN}@Office,su) exten =>
_4XX,4,Ha

[asterisk-users] Unable to find codec translation path with video enabled

[Administrator TOOTAI]

Hello list,

I want to connect 2 sites both having asterisk installed (1.4 and 
13.16from Ubuntu 14.04). When calling from 13.16 to 1.4 (call to echo 
test which should show video) I get in logs


[2017-06-13 14:45:26] WARNING[17176][C-03b0] channel.c: Unable to 
find a codec translation path: (h263|h264|h263p) -> (ulaw)
[2017-06-13 14:45:26] WARNING[17176][C-03b0] channel.c: Unable to 
find a codec translation path: (ulaw) -> (h263|h264|h263p)
[2017-06-13 14:45:26] VERBOSE[17140][C-03b0] app_dial.c: 
SIP/pubuntux-1acb answered SIP/197-1aca
[2017-06-13 14:45:26] VERBOSE[17140][C-03b0] chan_sip.c: Audio is at 
27374
[2017-06-13 14:45:26] VERBOSE[17140][C-03b0] chan_sip.c: Video is at 
XXX.XXX.XXX.XXX:28830
[2017-06-13 14:45:26] VERBOSE[17140][C-03b0] chan_sip.c: Adding 
codec ulaw to SDP
[2017-06-13 14:45:26] VERBOSE[17140][C-03b0] chan_sip.c: Adding 
video codec h263 to SDP
[2017-06-13 14:45:26] VERBOSE[17140][C-03b0] chan_sip.c: Adding 
video codec h264 to SDP
[2017-06-13 14:45:26] VERBOSE[17140][C-03b0] chan_sip.c: Adding 
non-codec 0x1 (telephone-event) to SDP


On the other end (1.4) I have

[Jun 13 14:45:24] VERBOSE[3671] logger.c: Capabilities: us - 0x3f1fff 
(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), 
peer - audio=0x38000c (ulaw|alaw|h263|h263p|h264)/video=0x38 
(h263|h263p|h264), combined - 0x38000c (ulaw|alaw|h263|h263p|h264)
[Jun 13 14:45:24] VERBOSE[3671] logger.c: Non-codec capabilities (dtmf): 
us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 
(telephone-event)
[Jun 13 14:45:24] DEBUG[3671] chan_sip.c: Our T38 capability = (0), peer 
T38 capability (0), joint T38 capability (0)
[Jun 13 14:45:24] VERBOSE[3671] logger.c: Peer audio RTP is at port 
XXX.XXX.XXX.XXX:26748
[Jun 13 14:45:24] VERBOSE[3671] logger.c: Peer video RTP is at port 
XXX.XXX.XXX.XXX:24552


There is no video on the phone, only audio. Calling the local echo test 
is OK as well as calling another local videophone. Connecting the phone 
to the 1.4 asterisk and dialing echo test gave me also the video as well 
as calling the remote echo test (the one from 13.16). In all cases I 
have ulaw|h263 as codecs which is correct


As you can see from above logs, everything seems to be correct in both 
directions, audio as well as video. The only problem is the codec 
translation path between audio and video codecs in channel.c


Does anyone know what does this message mean or what could be the problem?

Thanks for any hint

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Re: [asterisk-users] Working around missing libmyodbc in Debian Stretch

[Administrator TOOTAI]

Le 08/06/2017 à 15:15, J Montoya or A J Stiles a écrit :

On Thursday 08 Jun 2017, Olivier wrote:

Hello,

I'm building a new Asterisk system from source on Debian Stretch.
My building script fails as package libmyodbc is currently missing from
Debian Stretch repo.

Is there a work around this without leaving MySQL/MariaDB galaxy ?


This is why you should not use Debian testing for a server!  Testing is kept
"always installable" by the crude method of REMOVING broken packages until a
compatible version is ready.  This means you will occasionally find a package
with no install candidate.


Well, Stretch will be the new stable as of 17 june 2017 which for me is 
no more a testing version.


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Re: [asterisk-users] asterisk server - no sound

[Administrator TOOTAI]

Le 06/06/2017 à 16:25, Daniel Tryba a écrit :

On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote:

extensions.conf:
[home]
exten = 102,1,Answer()
same =  n,Wait(1)


If this is copy and paste, then your dialplan is broken (= should be =>)


Well, no. = or => are the same.

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Re: [asterisk-users] Automatically dial a number, then an extension

[Administrator TOOTAI]

Le 15/05/2017 à 17:34, Tech Support a écrit :

All;

I have an application that dials a list of numbers and then plays a
recorded message. My customer uses it to dial a list of customers to
confirm their appointment for the next day. No biggie, maybe 25 – 30
calls per day for customers who want the confirmation call. What they
need now is a way to dial an extension after the number is dialed and
answered. I’ve seen that before, but I just can't remember where. I was
wondering if anyone else has implemented something along these lines.
Any insight at all would be greatly appreciated.

Thanks Much;


Hi,

look at option G in dial command

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Re: [asterisk-users] moh reload not reloading/reading new musiconhold files

[Administrator TOOTAI]

Le 23/03/2017 à 20:17, Jonas Kellens a écrit :

Hello


is there any more information on how to reload/read musiconhold files ?


CLI> module reload res_musiconhold

--
Daniel


On 07-03-17 10:46, Jonas Kellens wrote:

Hello

I did not mention it but of course the MOH directory is listed in
/etc/asterisk/musiconhold.conf :

[default]
mode=files
directory=/var/lib/asterisk/moh

[myfolder_1]
mode=files
directory=/var/lib/asterisk/moh/myfolder/1
sort=alpha

[myfolder_2]
mode=files
directory=/var/lib/asterisk/moh/myfolder/2
sort=alpha

[myfolder_3]
mode=files
directory=/var/lib/asterisk/moh/myfolder/3
sort=alpha


No mather where I put the new file, it is never listed.

Untill a full restart of Asterisk ! Then it is listed. But is there no
other way to load/read a new MOH file than to completely restart
Asterisk ??


After Asterisk restart :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/macroform-the_simplicity
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
Class: myfolder_1
File: /var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity



Kind regards.




On 03-03-17 18:26, John Kiniston wrote:

Your new file is in the 'myfolder/1'' subdirectory of the MOH
directory.

Either move the file into the MOH directory or define a new class in
musiconhold.conf that is for your directory.


On Fri, Mar 3, 2017 at 7:19 AM, Jonas Kellens
mailto:jonas.kell...@telenet.be>> wrote:

Hello

using Asterisk 1.8.32.3

Current music on hold :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity

New musiconhold file :

[root@myserver ]# file
/var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav
/var/lib/asterisk/moh/myfolder/1/macroform-the_simplicity.wav:
RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit,
mono 8000 Hz

I issue a reload of the moh :

myserver*CLI> moh reload
myserver*CLI> module reload res_musiconhold.so
[Mar  3 15:04:53] -- Reloading module 'res_musiconhold.so'
(Music On Hold Resource)

Situation after reload :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity


Even a complete 'module reload' on Asterisk CLI does nothing :

myserver*CLI> module reload
[Mar  3 15:13:54]   == Parsing '/etc/asterisk/extconfig.conf':
[Mar  3 15:13:54]   == Found
[Mar  3 15:13:54]   == Parsing '/etc/asterisk/logger.conf': [Mar
3 15:13:54]   == Found
...
[Mar  3 15:13:54] -- Reloading module 'res_musiconhold.so'
(Music On Hold Resource)
...


Situation after reload :

myserver*CLI> moh show files
Class: default
File: /var/lib/asterisk/moh/macroform-robot_dity
File: /var/lib/asterisk/moh/macroform-cold_day
File: /var/lib/asterisk/moh/reno_project-system
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity



So, reloading musiconhold does not reload/read musiconhold files.


How to read/load new musiconhold files into asterisk ??


Kind regards.



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balance accounts, build a wall, set a bone, comfort the dying, take
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[asterisk-users] Some SIP and IAX Asterisk unreachable after server restart

[Administrator TOOTAI]

Hi all,

we have a running Asterisk 11.25.1 in a VM (qemu/kvm) OS being Debian 
7.11 (wheezy), the host OS being the same.


Problem: when we restart the server (eg host + VM), all customers 
Asterisk connecting without a VPN (doesn't matter which Asterisk 
version) are no more reachable. Same for IAX users.


On the host side, after restart, SIP packet are entering the host but 
NOT to the VM (tshark debug). In IAX, POKE packets sended by the VM are 
reaching the client Asterisk but no packet answer.


Everything is working again if we restart the customers Asterisk.

Any clue ?

Regards

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Re: [asterisk-users] Asterisk 11.24.0 Now Available

[Administrator TOOTAI]

Le 26/10/2016 à 12:21, Joshua Colp a écrit :

Administrator TOOTAI wrote:

Le 25/10/2016 à 23:10, Asterisk Development Team a écrit :

The Asterisk Development Team has announced the release of Asterisk
11.24.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.24.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!


[...]

Hello,

this is the output from 11.24.0 patch. I think that something is wrong.

root@pabx:/usr/src# egrep ^"\+\+\+" asterisk-11.24.0-patch
+++ b/.version
+++ b/ChangeLog
+++ /dev/null
+++ /dev/null
+++ b/asterisk-11.24.0-summary.html
+++ b/asterisk-11.24.0-summary.txt


Please file an issue on the issue tracker[1].

[1] https://issues.asterisk.org/jira



Done

https://issues.asterisk.org/jira/browse/ASTERISK-26502

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Re: [asterisk-users] Asterisk 11.24.0 Now Available

[Administrator TOOTAI]

Le 25/10/2016 à 23:10, Asterisk Development Team a écrit :

The Asterisk Development Team has announced the release of Asterisk 11.24.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.24.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!


[...]

Hello,

this is the output from 11.24.0 patch. I think that something is wrong.

root@pabx:/usr/src# egrep ^"\+\+\+" asterisk-11.24.0-patch
+++ b/.version
+++ b/ChangeLog
+++ /dev/null
+++ /dev/null
+++ b/asterisk-11.24.0-summary.html
+++ b/asterisk-11.24.0-summary.txt

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Re: [asterisk-users] IAX - Equivalent of SipAddHeader

[Administrator TOOTAI]

Le 24/10/2016 à 18:46, John Kiniston a écrit :

You can do it with IAXVAR.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_IAXVAR


exten => 100,1,Set(IAXVAR(myvar)=foo)
exten => 100,n,Dial(OTHERHOST/201)

on OTHERHOST
exten => 201,1,Set(myvar=${IAXVAR(myvar)})
exten => 201,n,NoOP(My variable is ${myvar})


Thanks for the information. As on voip-info.org this function is no more 
detailed and status is 1.2 I thought that it was removed from latest 
version.




On Mon, Oct 24, 2016 at 9:33 AM, Administrator TOOTAI mailto:ad...@tootai.net>> wrote:

Hi list,

is there any existing IAX command to add information to a call like
SipAddHeader? Another solution is sending text frame (0x07) frame
type, but I don know how do it in a dialplan.

Thanks for any hint.

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[asterisk-users] IAX - Equivalent of SipAddHeader

[Administrator TOOTAI]

Hi list,

is there any existing IAX command to add information to a call like 
SipAddHeader? Another solution is sending text frame (0x07) frame type, 
but I don know how do it in a dialplan.


Thanks for any hint.

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Re: [asterisk-users] iptables for SIP talk to other port

[Administrator TOOTAI]

Le 15/10/2016 à 18:17, Jerry Geis a écrit :

I have a host 192.168.1.3 that wants to run SIP on 5068 (long story).
My host is 192.168.10.201.
My host needs to stay on 5060 because of all the other devices I have
connected.

I tried putting port=5068 in my SIP extension definition but that did
not work.

So I thought about using iptables to accomplish this:

iptables -t nat -A PREROUTING  -p tcp --dport 5068-j
REDIRECT --to-port 5060
iptables -t nat -A POSTROUTING -p tcp --dport 5060 -d 192.168.1.3 -j
REDIRECT --to-port 5068


Do I not have the right format of the command?
Anything incoming destined for 5068 redirect to 5060...
Anything going out to 192.168.1.3 and port 5060 redirect to 5068.

Seems like that should have worked?

Thoughts?  sip show peers still says unreachable.


Generally SIP is UDP not TCP. Did you modify your asterisk.conf to TCP ?

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Re: [asterisk-users] Asterisk 13 PJSIP with Snom 710

[Administrator TOOTAI]

Le 09/09/2016 à 18:32, Madushan Geethanga a écrit :

Hi,


If you're not using RTP encryption did you uncheck the option in your 
RTP TAB from identity ?




This is the log. ex dialling 0 from snom phone


<--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878
 --->
INVITE sip:0@54.206.59.252 ;user=phone SIP/2.0
Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
From: "outburns00-nhvg5vjjn6-2001"
mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252>>;tag=1bb809zgaa
To: mailto:sip%3A0@54.206.59.252>;user=phone>
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 INVITE
Max-Forwards: 70
User-Agent: snom710/8.7.5.35 
Contact: http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835>>;reg-id=1
X-Serialnumber: 000413747C96
P-Key-Flags: resolution="31x13", keys="4"
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600
Min-SE: 90
Content-Type: application/sdp
Content-Length: 405

v=0
o=root 2136927789 2136927789 IN IP4 192.168.2.28
s=call
c=IN IP4 123.231.72.210
t=0 0
m=audio 62724 RTP/AVP 9 0 8 3 99 112 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (572 bytes) to UDP:123.231.72.210:33878
 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
123.231.72.210:45835;rport=33878;received=123.231.72.210;branch=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
From: "outburns00-nhvg5vjjn6-2001"
mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252>>;tag=1bb809zgaa
To: mailto:sip%3A0@54.206.59.252>;user=phone>;tag=z9hG4bK-bskkkx1t5bas
CSeq: 1 INVITE
WWW-Authenticate: Digest
realm="asterisk",nonce="1473438693/ef923d25464dbedc1dbd85e0ccea08b7",opaque="210b270d7abb2354",algorithm=md5,qop="auth"
Server: Asterisk PBX certified/13.8-cert2
Content-Length:  0


<--- Received SIP request (487 bytes) from UDP:123.231.72.210:33878
 --->
ACK sip:0@54.206.59.252 ;user=phone SIP/2.0
Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport
From: "outburns00-nhvg5vjjn6-2001"
mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252>>;tag=1bb809zgaa
To: mailto:sip%3A0@54.206.59.252>;user=phone>;tag=z9hG4bK-bskkkx1t5bas
Call-ID: 313437333433383639323238313539-ahn3begiq66q
CSeq: 1 ACK
Max-Forwards: 70
User-Agent: snom710/8.7.5.35 
Contact: http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835>>;reg-id=1
Content-Length: 0


Best Regards,
Madushan



On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga
mailto:mgliyanage...@gmail.com>> wrote:

Hi,

I'm trying to setup snom 710 phone with asterisk 13 with PJSIP.
inbound is working fine but i cannot dial out. i don't hear anything
on the phone and asterisk CLI also does not show anything. my config
is. please advice.

[2001]
type=endpoint
context=out-local
disallow=all
allow=ulaw
allow=alaw
transport=system-udp
auth=2001
aors=2001
direct_media=no
rtp_symmetric=yes
force_rport=yes
allow=alaw
allow=speex
allow=speex16
allow=speex32
allow=gsm


[2001]
type=aor
qualify_frequency=5000
authenticate_qualify=yes
max_contacts=1
remove_existing=yes

[2001]
type=auth
auth_type=userpass
password=test
username=test

Best Regards,
Madushan






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Re: [asterisk-users] Trouble getting peer variable (sip username) on 302 Moved Temporarily

[Administrator TOOTAI]

Le 02/09/2016 à 11:26, Jonas Kellens a écrit :

Hello

when setting a local forward (in this case to extension 23) on a SIP
phone, I see the following on the Asterisk CLI :


[Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back
from 11.22.33.44:40670
[Aug 31 14:59:34] -- Now forwarding
Local/myaccount184@CallFromQueue-07f4;2 to 'Local/23@from-internal'
(thanks to SIP/myaccount184-3729)


Question : how can I read the variable which contains the value
'myaccount184' in the context from-internal ?


From SIP_HEADER(TO) ?

[...]

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Re: [asterisk-users] Multiple phones when one is unregistered

[Administrator TOOTAI]

Le 01/09/2016 à 17:27, D'Arcy J.M. Cain a écrit :

On Thu, 1 Sep 2016 13:49:57 + (UTC)
t...@softins.co.uk (Tony Mountifield) wrote:

What module am I missing?


The ExecIf command is provided in the module app_exec, which is
usually located at /usr/lib/asterisk/modules/app_exec.so


Yes, I see it.


Maybe you had turned off app_exec in the menuconfigi when building,
or maybe your modules.conf has a noload => app_exec.so


The distributed modules.conf does not appear to mention it at all.  I
don't need it now after all so I won't add it in until I can evaluate
the security issues that it might bring with it.  It's hard to find
documentation for it other than the actual source code.

http://doxygen.asterisk.org/trunk/dc/d73/app__exec_8c-source.html



You can do it with GotoIf
exten => 55,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,GotoIf($["${DEVICE_STATE(SIP/user1)}" != "NOT IN USE"]?User2)
same => n,Set(toRing=${toRing}&SIP/user1)
same => n(User2),GotoIf($["${DEVICE_STATE(SIP/user2)}" != "NOT IN 
USE"]?User3)

same => n,Set(toRing=${toRing}&SIP/user3)
same => n(User3),GotoIf($["${DEVICE_STATE(SIP/user3)}" != "NOT IN 
USE"]?Call)

same => n,Set(toRing=${toRing}&SIP/user3)
same => n(Call),GotoIf($["x${toRing}" = "x"]?NoPhoneToCall)
same => n,Dial(${toRing:1}) ;to remove the first &

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Re: [asterisk-users] Multiple phones when one is unregistered

[Administrator TOOTAI]

Le 01/09/2016 à 03:57, D'Arcy J.M. Cain a écrit :

On Tue, 30 Aug 2016 17:56:35 +0200
Administrator TOOTAI  wrote:

Something like

exten => 55,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user1)


Failed.  I checked the online docs and the syntax seems to be correct


No. The trailing ) is missing


but I get this:

[Aug 31 21:52:00] WARNING[-1][C-0001fed5] pbx.c: No application
'ExecIf' for extension (unauthenticated, 55, 3)

Is there a module that I need to load?

In case it matters I am running Asterisk 11.23.0 on NetBSD 7.0.


What's the output of CLI command "core show application ExecIf" ?

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Re: [asterisk-users] Multiple phones when one is unregistered

[Administrator TOOTAI]

Le 30/08/2016 à 18:05, D'Arcy J.M. Cain a écrit :

On Tue, 30 Aug 2016 17:56:35 +0200
Administrator TOOTAI  wrote:

exten => 55,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user1)
same => n,ExecIf($["${DEVICE_STATE(SIP/user2)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user2)
same => n,ExecIf($["${DEVICE_STATE(SIP/user3)}" = "NOT IN
USE"]?Set(toRing=${toRing}&SIP/user3)
same => n,Dial(${toRing:1}) ;to remove the first &

would do the work


That looks good and is easy to add and delete from the list.  I will
give this a try one night this week.  Not sure what that last line
would do if all of the phones are off but if they are the buzzer won't
be answered anyway.


Don't execute the Dial cmd if ${toRing} is empty ;-)
 ...
 same => n,ExecIf($["x${toRing}" != "x"]?Dial(${toRing:1})) ;to remove 
the first &


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Re: [asterisk-users] Multiple phones when one is unregistered

[Administrator TOOTAI]

Le 30/08/2016 à 15:56, D'Arcy J.M. Cain a écrit :

I have an extension that looks like this:

exten => 55,1,Verbose(Door buzzer calling)
  same => n,Dial(SIP/user1&SIP/user2&SIP/user3)

The idea is that any of the three users can answer the phone to let
someone in.  The problem is that if, say, user2 unplugs his phone then
the call immediately goes to his voice mail and the other two do not
have the ability to open the door.

Is there any way to direct only to phones in a list that are currently
registered?  I am sure that I can write a rather convoluted extension
to check for registrations and create a dial command but I am hoping
that there is an easier way so that I can create these types of
extensions for other clients easily as well as being able to add and
remove destinations quickly.


Something like

exten => 55,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN 
USE"]?Set(toRing=${toRing}&SIP/user1)
same => n,ExecIf($["${DEVICE_STATE(SIP/user2)}" = "NOT IN 
USE"]?Set(toRing=${toRing}&SIP/user2)
same => n,ExecIf($["${DEVICE_STATE(SIP/user3)}" = "NOT IN 
USE"]?Set(toRing=${toRing}&SIP/user3)

same => n,Dial(${toRing:1}) ;to remove the first &

would do the work
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Re: [asterisk-users] my dahdi dont'n start

[Administrator TOOTAI]

Le 26/04/2016 17:23, Mamadou NGOM a écrit :

Hello,


Having installed DAHDI to be able to use the meetme() application , when
I start the dahdi service it generates me the following error:

-bash: /etc/init.d/dahdi: No such file or directory


Clear, the file dahdi is not existing. Did you copy it?

BTW, you shouldn't need dahdi to run meetme. BTW #2, depending on your 
asterisk version, meetme is replaced by ConfBridge


[...]

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Re: [asterisk-users] Incoming INVITE with Portability Info and LRN

[Administrator TOOTAI]

Le 18/03/2016 16:20, Trey Hilyard a écrit :

I am trying to set up my Asterisk server so that it will recognize an
incoming call to the Asterisk's own Location Routing Number (LRN),
validating the "rn" in the INVITE and then using the Called Number from
the INVITE as the extension in the dialplan.

The INVITE R-URI looks like:
INVITE
sip:+19135041291;rn=+1913663;npdi@12.4.240.200:5060;user=phone;transport=udp
SIP/2.0

The +1913663000 is the LRN of the Asterisk box, so I would want to have
the dialplan validate that the "rn" is that number. The +19136631291 is
the extension within the system that they are trying to reach, that
extension will vary, and will have an exten defined in the dialplan.

I assume that this is just going to require that I do some matching and
substring-type variable replacement to hit a context with just the
Called Number part of the request, but I wondered if anyone had a
working example of this before I started putting too much effort into it.


Use the SIP_HEADER function

http://www.voip-info.org/wiki/view/Asterisk+func+sip_header

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Re: [asterisk-users] Fwd: Unable to place outbound calls

[Administrator TOOTAI]

Le 15/03/2016 11:20, Feroz Ahmed a écrit :

Hi I need help


Hello Ahmed



This is the error:



[...]


[Mar 14 19:55:15] WARNING[20595][C-000b]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Auto fallthrough, channel 'SIP/1001-000b' status is
'CHANUNAVAIL'


CHANUNAVAIL and here is why


sonetel/feroz.sonetel 212.72.62.126D
  Auto (No)  Yes5060 UNREACHABLE


Unreachable, you face a network problem.

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Re: [asterisk-users] variables or including other files in followme.conf

[Administrator TOOTAI]

Le 10/03/2016 15:12, Karl Anderson a écrit :

Is is possible to use variables in followme.conf? Is it possible to
include another conf file?




in the followme.conf file

#include local/followme.d/*.conf

will include all files having .conf extension in this directory

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Re: [asterisk-users] [dongle0] timedout while waiting 'OK' in response to 'AT'

[Administrator TOOTAI]

Hi Vitor

Le 12/02/2016 19:17, Vitor Mazuco a écrit :

I think that my monden is locked for Voice

I use a Huawei E173, someone know how can I unlock it?

Is necessary to upgrade the firmware?


Google is your friend, plenty of informations like

http://www.modemunlock.com/huawei-e173-unlock-3g-usb-modem.html

Regards

Daniel




2016-02-12 15:39 GMT-02:00, Vitor Mazuco :

I tried this

[dongle0]
;audio=/dev/ttyUSB1 ; tty port for audio connection;
  no default value
;data=/dev/ttyUSB2  ; tty port for AT commands;
  no default value

; or you can omit both audio and data together and use
imei=123456789012345 and/or imsi=123456789012345
;  imei and imsi must contain exactly 15 digits !
;  imei/imsi discovery is available on Linux only
imei=352098043831724
;imsi=123456789012345


My imei is 352098043831724

But nothing change.

2016-02-12 15:12 GMT-02:00, Frank :

On Fri, 2016-02-12 at 14:33 -0200, Vitor Mazuco wrote:

Yes I used.

The problem can be the version of Asterisk?

I use Asterisk 13 instead of 11.


Try

[dongle0]
imei=347654458453667
imsi=976895757545778



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[asterisk-users] Asterisk 13.7.0 failed to start - PJSIP 2.4.5

[Administrator TOOTAI]

Hello,

We installed the subject detailed versions on a uptodate debian wheezy. 
When starting Asterisk we get


 Loading chan_pjsip.so.
  == Registered RTP glue 'PJSIP'
  == Registered channel type 'PJSIP' (PJSIP Channel Driver)
18:26:10.812 sip_endpoint.c !Module "mod-refer" registered
asterisk: ../src/pjsip-simple/evsub.c:417: pjsip_evsub_register_pkg: 
Assertion `mod_evsub.mod.id != -1' failed.


Any clue on what could be the problem ?

Thanks

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Re: [asterisk-users] how to get an info from "To:" header?

[Administrator TOOTAI]

Le 17/09/2015 12:37, Дорофеев Сергей a écrit :

Hello list!


Hello



Sorry for kinda dumb question, I guess, but I have too little time to
research it by myself.

I have a SIP packet, which looks like this:

<--- SIP read from UDP:10.186.0.38:5060 --->

INVITE sip:XXX@10.186.35.98:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.186.0.38:5060;branch=z9hG4bKh4utm43008vheqk093b0.1

Call-ID: ba9vp4zsbbsfi0vagdafg0vpzpp0z9wh@SoftX3000

From: ;tag=zwbzfehp-CC-22

To: 

CSeq: 1 INVITE

Contact: 

Min-SE: 90

Session-Expires: 300

Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER

User-Agent: Huawei SoftX3000 V300R011

Diversion:
;reason=unconditional;counter=1

Supported: 100rel,timer

Max-Forwards: 69

Content-Length: 338

Content-Type: application/sdp

Priority: urgent

I need to use info from fields “To:” and “Contact:” later in my
dialplan. I belive, I have to do something like “exten =>
_/X.,1,Set(VAR=${WHAT/_SHOULD_I_TYPE_HERE?})”


Sample:

exten => s,1,Set(__DIALEDNUMBER=${SIP_HEADER(TO):5})
exten => s,n,Set(__DIALEDNUMBER=${CUT(DIALEDNUMBER,@,1)})
...

Regards

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Re: [asterisk-users] Asterisk 11.19.0 Now Available

[Administrator TOOTAI]

Le 07/08/2015 23:54, Asterisk Development Team a écrit :

The Asterisk Development Team has announced the release of Asterisk 11.19.0.

[...]

Hello,

We have problem with patches since 11.18.0 We have to download the full 
tar.gz to get last version :-(.


Before this, since ages, we used to patch the previous version like

#patch -p0 < ../asterisk-11.17.0-patch

(applied to the current asterisk-11.16-0 directory), compile and 
install. That's all, servers where uptodate, job done.


Taking a look at the header from asterisk-11.17.0-patch (and previous) 
we see


--- asterisk-11.16.0-summary.html  (.../11.16.0)   (revision 433916)
+++ asterisk-11.16.0-summary.html  (.../11.17.0)   (revision 433916)

which is, diff between asterisk-11.16.0 and -in this case- the new 
asterisk-11.17.0


Now, since 11.18.0 version, patch is looking like

diff --git a/.version b/.version
index cde331b..3644f46 100644
--- a/.version
+++ b/.version
@@ -1 +1 @@
-11.19.0-rc1
\ No newline at end of file
+11.19.0 



\ No newline at end of file

As you can see patch is build against 11.19.0-rc1, not 11.18.0

How can we apply this patch to a legacy asterisk-11.18.0 tar.gz ?

Thanks for any hint.

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Re: [asterisk-users] 11.18.0 patch against 11.17.0 running version failed to apply

[Administrator TOOTAI]

Le 09/07/2015 17:05, Tzafrir Cohen a écrit :

On Thu, Jul 09, 2015 at 12:28:15AM +0200, Administrator TOOTAI wrote:


zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p1 <
../asterisk-11.18.0-patch
patching file .version
Hunk #1 FAILED at 1.
1 out of 1 hunk FAILED -- saving rejects to file .version.rej
patching file ChangeLog
Hunk #1 FAILED at 1.
1 out of 1 hunk FAILED -- saving rejects to file ChangeLog.rej
The next patch would delete the file asterisk-11.18.0-rc1-summary.html,
which does not exist!  Assume -R? [n]
Apply anyway? [n]
Skipping patch.
1 out of 1 hunk ignored
The next patch would delete the file asterisk-11.18.0-rc1-summary.txt,
which does not exist!  Assume -R? [n]
Apply anyway? [n]
Skipping patch.
1 out of 1 hunk ignored
patching file asterisk-11.18.0-summary.html
patching file asterisk-11.18.0-summary.txt

As you can see, patch is against -rc1 not 11.17.0 ...


The content of files has changed. patch refuses to change from an
unfamiliar content.

Either edit the patch file and remove .version (edit the version
manually) or edit the patch file and edit the version form 11.7.0 to
11.8.0 .

The content of the files you refer to is normally insignificant to the
behaviour of Asterisk. Just remove them from the patch and be done with
it.



Well take a look in the file, this patch is only for:

--- a/.version
+++ b/.version

--- a/ChangeLog
+++ b/ChangeLog

--- a/asterisk-11.18.0-rc1-summary.html
+++ /dev/null

--- a/asterisk-11.18.0-rc1-summary.txt
+++ /dev/null

--- /dev/null
+++ b/asterisk-11.18.0-summary.txt

Definitely something is wrong with this patch.

We reinstalled our servers from full tar.gz

Thanks for your support

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Re: [asterisk-users] 11.18.0 patch against 11.17.0 running version failed to apply

[Administrator TOOTAI]

Le 08/07/2015 17:36, Richard Mudgett a écrit :



On Wed, Jul 8, 2015 at 8:14 AM, Administrator TOOTAI mailto:ad...@tootai.net>> wrote:

Hi list,

we wanted to patch our servers with 11.18.0 patch against 11.17.0
actual running version. Patch failed with

zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p0 <
../asterisk-11.18.0-patch
can't find file to patch at input line 5
Perhaps you used the wrong -p or --strip option?
The text leading up to this was:
--
|diff --git a/.version b/.version
|index c5df2aa..150754a 100644
|--- a/.version
|+++ b/.version
--
File to patch:


[SNIP]



The two patch files were created by different version control systems.
One was created by git
the other created by subversion.  For the git patch you would need to
use -p1 for the subversion
patch you would need to use -p0.  The patch program gave you this hint
when it failed to apply
the patch: "Perhaps you used the wrong -p or --strip option?".


We already tried with no luck:

zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p1 < 
../asterisk-11.18.0-patch

patching file .version
Hunk #1 FAILED at 1.
1 out of 1 hunk FAILED -- saving rejects to file .version.rej
patching file ChangeLog
Hunk #1 FAILED at 1.
1 out of 1 hunk FAILED -- saving rejects to file ChangeLog.rej
The next patch would delete the file asterisk-11.18.0-rc1-summary.html,
which does not exist!  Assume -R? [n]
Apply anyway? [n]
Skipping patch.
1 out of 1 hunk ignored
The next patch would delete the file asterisk-11.18.0-rc1-summary.txt,
which does not exist!  Assume -R? [n]
Apply anyway? [n]
Skipping patch.
1 out of 1 hunk ignored
patching file asterisk-11.18.0-summary.html
patching file asterisk-11.18.0-summary.txt

As you can see, patch is against -rc1 not 11.17.0 ...

Thanks for your support.

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[asterisk-users] 11.18.0 patch against 11.17.0 running version failed to apply

[Administrator TOOTAI]

Hi list,

we wanted to patch our servers with 11.18.0 patch against 11.17.0 actual 
running version. Patch failed with


zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p0 < 
../asterisk-11.18.0-patch

can't find file to patch at input line 5
Perhaps you used the wrong -p or --strip option?
The text leading up to this was:
--
|diff --git a/.version b/.version
|index c5df2aa..150754a 100644
|--- a/.version
|+++ b/.version
--
File to patch:

It seems that patch file for 11.18.0 are completely different from 
previous one. Patch for 11.17.0 looked like (first 3 lines)


--- asterisk-11.16.0-summary.html   (.../11.16.0) (revision 433916)
+++ asterisk-11.16.0-summary.html   (.../11.17.0) (revision 433916)
@@ -1,307 +0,0 @@

which is different from 11.18.0

diff --git a/.version b/.version
index c5df2aa..150754a 100644
--- a/.version
+++ b/.version
@@ -1 +1 @@

OS is Debian Wheezy 7.8

What are we doing wrong ?

Thanks for any hint

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[asterisk-users] 11.18.0 patch mistake

[Administrator TOOTAI]

Hi,

we think that there is a mistake with the asterisk-11.18.0.patch. The 
file look like


diff --git a/.version b/.version
index c5df2aa..150754a 100644
--- a/.version
+++ b/.version
@@ -1 +1 @@
-11.18.0-rc1
\ No newline at end of file
+11.18.0 



\ No newline at end of file
[...]

trying to apply the patch fail with error

can't find file to patch at input line 5
Perhaps you used the wrong -p or --strip option?
The text leading up to this was:
--
|diff --git a/.version b/.version
|index c5df2aa..150754a 100644
|--- a/.version
|+++ b/.version
--

Regards

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Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In

[Administrator TOOTAI]

Le 01/05/2015 00:05, Andrew Martin a écrit :

- Original Message -

From: "Administrator TOOTAI" 
To: asterisk-users@lists.digium.com
Sent: Thursday, April 30, 2015 4:43:33 PM
Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In


I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and
internal phones are located on the 10.10.32.0/21 LAN subnet. I have many
internal SIP phones, which appear to be working correctly. I have a few
external phones (Yealink SIP-T32G or other Yealink model) on
192.168.32.0/24 which have an OpenVPN client configured on them that
connects back to the LAN network through a pfSense gateway with OpenVPN
configured on it.


I faced problems with pfsense -no VPN involved- and finally installed
siproxd on it. Also set the firewall mode to conservative.


Daniel,

Thanks for the information. Do you have an example or documentation on the
siproxd configuration that you used?


No, just follow the basis of the parameters given by the package. If I 
remember, SIP use the proxy siproxd and RTP is direct.


Another solution I used on an not stable xDSL line, was to install 
asterisk on pfsense, this asterisk taking only care on the local traffic 
(call from local extension to local extension). The asterisk register 
with the main one as a trunk for incoming/outgoing calls. Worked too.


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Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In

[Administrator TOOTAI]

Le 30/04/2015 19:18, Andrew Martin a écrit :

Hello,


Hello



I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal 
phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP 
phones, which appear to be working correctly. I have a few external phones 
(Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an 
OpenVPN client configured on them that connects back to the LAN network through 
a pfSense gateway with OpenVPN configured on it.


I faced problems with pfsense -no VPN involved- and finally installed 
siproxd on it. Also set the firewall mode to conservative.


[...]

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[asterisk-users] Sending DTMF on not answered channel

[Administrator TOOTAI]

Hello,

I setup a door open system with a basic DTMF card. The card is connected 
to an Sipura/Linksys 3102 FXS port and is powered by this port.


My problem is that when I send a call with Dial() command, channel has 
to be answered before receiving DTMFs, what my card does not. Is there a 
way to autoanswer those type of calls or to send DTMF on an non answered 
channel or another solution/idea to rich the goal ?


Only solution I found is to connect an analog phone in parallel to my 
card and to set it in autoanswer mode: at this moment the card get the 
DTMF sended via SendDTMF() command.


Thanks for any hint. Asterisk is 11.15.0 from Elastix 2.5

Regards

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[asterisk-users] Question about hangup - Asterisk v11.15.0

[Administrator TOOTAI]

Hello,

on previous versions of asterisk, extension h and H make us know who 
ended a call (caller or callee). In the last * versions, seems that only 
h extension is used, as stated here 
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions


In the last versions, how do we know which end terminate a call (SIP, 
ISDN, Analog, ...) in h extension ? Will the 
${HASH(SIP_CAUSE,${CDR(dstchannel)})} give the information ?


We also face a strange behavior: we are ringing few phones (~10) and 
sometimes, once the call get answered, we see that 2~3 seconds after 
this, music on hold is started on the channel! And 20 seconds after, the 
call is terminated without that any party hanged up :-(


It's a Elastix 2.5 installation, we thought that problem could came from 
Elastix so we set our own dialplan for incoming calls:


 same = 
n,Set(__phonesToRing=SIP/118&SIP/119&SIP/122&SIP/123&SIP/124&SIP/125&SIP/126&SIP/127&SIP/128&SIP/129&SIP/130&SIP/132)

 same = n(startRing),Answer()
 same = n,Dial(${phonesToRing},,it) ;no voicemail 
or forward => ring indefenitely

 same = n,Hangup

Incoming call give for instance in logs:

[2015-03-23 11:07:20] VERBOSE[1342][C-0e85] app_dial.c: -- 
SIP/126-43d8 is ringing
[2015-03-23 11:07:21] VERBOSE[1342][C-0e85] app_dial.c: -- 
SIP/118-43d3 connected line has changed. Saving it until answer for 
SIP/bero_trunk-43d2
[2015-03-23 11:07:21] VERBOSE[1342][C-0e85] app_dial.c: -- 
SIP/118-43d3 answered SIP/bero_trunk-43d2
[2015-03-23 11:07:25] VERBOSE[1342][C-0e85] res_musiconhold.c: 
-- Started music on hold, class 'default', on SIP/bero_trunk-43d2
[2015-03-23 11:07:27] VERBOSE[1342][C-0e85] res_musiconhold.c: 
-- Stopped music on hold on SIP/bero_trunk-43d2
[2015-03-23 11:07:41] VERBOSE[1342][C-0e85] pbx.c: -- Executing 
[h@from-trunk:1] Macro("SIP/bero_trunk-43d2", "hangupcall,") in new 
stack


Thanks for any hint

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Re: [asterisk-users] chanspy for group extension

[Administrator TOOTAI]

Hi,

Le 12/03/2015 17:28, Salaheddine Elharit a écrit :

hello list,

i use the code below

[macro-chanspy]
exten => s,1,Authenticate(${ARG1})
exten => s,n,ChanSpy(SIP/${EXTEN:3},__dqs)


Here you have a problem: ${EXTEN} value is s

[...]

Daniel

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Re: [asterisk-users] Res_fax - FAXOPT(faxdetect)

[Administrator TOOTAI]

Le 18/02/2015 18:52, Eric Wieling a écrit :


I solved the issue by not answering the call as I assume others have done.


The only solution we found is to set faxdetect=no in the sip.conf of the 
peer definition. The Set(FAXOPT(detect)=[yes|no]) command in the 
dialplan is not taken in account.


The problem with this solution is that we can't mix fax reception with 
direct fax line and fax detection on audio line. Or better said, to use 
a mix of those detection, we should put a Wait(x) in the fax direct DID 
and systematically use the fax extension to dial hylafax like


[FAXDirectDID]

exten = fax,1,Dial(IAX2/300,,)
 same = n,Hangup

exten = _X.,1,Wait(10)
 same = n,Congestion()

Not particulary clean.

Daniel



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator 
TOOTAI
Sent: Wednesday, February 18, 2015 12:50 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Res_fax - FAXOPT(faxdetect)


Hello


Le 17/02/2015 17:00, Administrator TOOTAI a écrit :

Hi,

as stated in the documentation, it's allowed to set
FAXOPT(faxdetect)=yes/no to allow fax detection.

It's done (see below) but still fax detection :-( Extension 300 is
hylafax with iaxmodem.

On the upper Asterisk gw it's the same, despite the faxdetect set to no
we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile
phone calling the 0123456789 PSTN number.

  -- Executing [0123456789@from-internal:1]
Set("SIP/TOOTAi-8262", "FAXOPT(faxdetect)=no") in new stack
  -- Executing [0123456789@from-internal:2]
Macro("SIP/TOOTAi-8262", "Fax") in new stack
  -- Executing [s@macro-Fax:1] Dial("SIP/TOOTAi-8262",
"IAX2/300,,") in new stack
  -- Called IAX2/300
  -- Call accepted by 127.0.0.1 (format alaw)
  -- Format for call is (alaw)
  -- IAX2/300-7211 is ringing
  -- IAX2/300-7211 answered SIP/TOOTAiAudio-8262
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
[2015-02-17 16:52:51] NOTICE[3467][C-1d5b]: chan_sip.c:10645
process_sdp: T.38 re-INVITE detected but no fax extension
[2015-02-17 16:52:56] WARNING[3467][C-1d5b]: chan_sip.c:9868
process_sdp: Insufficient information for SDP (m= not found)
  -- Executing [h@from-internal:1] Hangup("SIP/TOOTAi-8262", "")
in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/TOOTAi-8262'
  -- Hungup 'IAX2/300-7211'

Thanks for your support



No one have an idea on this ?



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Re: [asterisk-users] Res_fax - FAXOPT(faxdetect)

[Administrator TOOTAI]

Le 18/02/2015 18:52, Eric Wieling a écrit :


I solved the issue by not answering the call as I assume others have done.


That's my problem: call is NOT answered :-( or better said, is answered 
by hylafax. That's why I thought that setting faxopt(faxdetect)=no would 
put asterisk out of the path.


Asterisk version is 11.15.0 from Elastix. Same happend on a stock 11.16.0

Thanks for your answer



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator 
TOOTAI
Sent: Wednesday, February 18, 2015 12:50 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Res_fax - FAXOPT(faxdetect)


Hello


Le 17/02/2015 17:00, Administrator TOOTAI a écrit :

Hi,

as stated in the documentation, it's allowed to set
FAXOPT(faxdetect)=yes/no to allow fax detection.

It's done (see below) but still fax detection :-( Extension 300 is
hylafax with iaxmodem.

On the upper Asterisk gw it's the same, despite the faxdetect set to no
we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile
phone calling the 0123456789 PSTN number.

  -- Executing [0123456789@from-internal:1]
Set("SIP/TOOTAi-8262", "FAXOPT(faxdetect)=no") in new stack
  -- Executing [0123456789@from-internal:2]
Macro("SIP/TOOTAi-8262", "Fax") in new stack
  -- Executing [s@macro-Fax:1] Dial("SIP/TOOTAi-8262",
"IAX2/300,,") in new stack
  -- Called IAX2/300
  -- Call accepted by 127.0.0.1 (format alaw)
  -- Format for call is (alaw)
  -- IAX2/300-7211 is ringing
  -- IAX2/300-7211 answered SIP/TOOTAiAudio-8262
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
[2015-02-17 16:52:51] NOTICE[3467][C-1d5b]: chan_sip.c:10645
process_sdp: T.38 re-INVITE detected but no fax extension
[2015-02-17 16:52:56] WARNING[3467][C-1d5b]: chan_sip.c:9868
process_sdp: Insufficient information for SDP (m= not found)
  -- Executing [h@from-internal:1] Hangup("SIP/TOOTAi-8262", "")
in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/TOOTAi-8262'
  -- Hungup 'IAX2/300-7211'

Thanks for your support



No one have an idea on this ?



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Re: [asterisk-users] Res_fax - FAXOPT(faxdetect)

[Administrator TOOTAI]

Hello


Le 17/02/2015 17:00, Administrator TOOTAI a écrit :

Hi,

as stated in the documentation, it's allowed to set
FAXOPT(faxdetect)=yes/no to allow fax detection.

It's done (see below) but still fax detection :-( Extension 300 is
hylafax with iaxmodem.

On the upper Asterisk gw it's the same, despite the faxdetect set to no
we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile
phone calling the 0123456789 PSTN number.

 -- Executing [0123456789@from-internal:1]
Set("SIP/TOOTAi-8262", "FAXOPT(faxdetect)=no") in new stack
 -- Executing [0123456789@from-internal:2]
Macro("SIP/TOOTAi-8262", "Fax") in new stack
 -- Executing [s@macro-Fax:1] Dial("SIP/TOOTAi-8262",
"IAX2/300,,") in new stack
 -- Called IAX2/300
 -- Call accepted by 127.0.0.1 (format alaw)
 -- Format for call is (alaw)
 -- IAX2/300-7211 is ringing
 -- IAX2/300-7211 answered SIP/TOOTAiAudio-8262
   == Using UDPTL TOS bits 184
   == Using UDPTL CoS mark 5
[2015-02-17 16:52:51] NOTICE[3467][C-1d5b]: chan_sip.c:10645
process_sdp: T.38 re-INVITE detected but no fax extension
[2015-02-17 16:52:56] WARNING[3467][C-1d5b]: chan_sip.c:9868
process_sdp: Insufficient information for SDP (m= not found)
 -- Executing [h@from-internal:1] Hangup("SIP/TOOTAi-8262", "")
in new stack
   == Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/TOOTAi-8262'
 -- Hungup 'IAX2/300-7211'

Thanks for your support



No one have an idea on this ?

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[asterisk-users] Res_fax - FAXOPT(faxdetect)

[Administrator TOOTAI]

Hi,

as stated in the documentation, it's allowed to set 
FAXOPT(faxdetect)=yes/no to allow fax detection.


It's done (see below) but still fax detection :-( Extension 300 is 
hylafax with iaxmodem.


On the upper Asterisk gw it's the same, despite the faxdetect set to no 
we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile 
phone calling the 0123456789 PSTN number.


-- Executing [0123456789@from-internal:1] 
Set("SIP/TOOTAi-8262", "FAXOPT(faxdetect)=no") in new stack
-- Executing [0123456789@from-internal:2] 
Macro("SIP/TOOTAi-8262", "Fax") in new stack
-- Executing [s@macro-Fax:1] Dial("SIP/TOOTAi-8262", 
"IAX2/300,,") in new stack

-- Called IAX2/300
-- Call accepted by 127.0.0.1 (format alaw)
-- Format for call is (alaw)
-- IAX2/300-7211 is ringing
-- IAX2/300-7211 answered SIP/TOOTAiAudio-8262
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
[2015-02-17 16:52:51] NOTICE[3467][C-1d5b]: chan_sip.c:10645 
process_sdp: T.38 re-INVITE detected but no fax extension
[2015-02-17 16:52:56] WARNING[3467][C-1d5b]: chan_sip.c:9868 
process_sdp: Insufficient information for SDP (m= not found)
-- Executing [h@from-internal:1] Hangup("SIP/TOOTAi-8262", "") 
in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 
'SIP/TOOTAi-8262'

-- Hungup 'IAX2/300-7211'

Thanks for your support

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[asterisk-users] Deactivate faxdetect on IAX channel

[Administrator TOOTAI]

Hi,

we have following setup:

fax machines > PSTN -> GW SIP -> Asterisk -> Peer IAX (Elastix)

Asterisk is 11.16 as well as Asterisk version of Elastix peer.

When sending incoming fax calls to the Peer IAX -which receive them 
using hylafax- we want to tell our asterisk to NOT detect fax CNG. It's 
easy on a SIP channel as faxdetect=no can be set for the peer, but how 
to do it for an IAX peer? Is this option available too?


Also, don't answering the call before sending it to the IAX peer is not 
an option.


Thanks for any hint

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Re: [asterisk-users] Investigating international calls fraud

[Administrator TOOTAI]

Le 28/01/2015 22:03, Steven McCann a écrit :

Hello,


Hi



I'm investigating a situation where there was a hundreds of minutes of
calls from an internal SIP extension to an 855 number in Cambodia,
resulting in a crazy ($25,000+) bill from the phone company. I'm
investigating, but can anyone provide some feedback on what's happened
here? I'm investigating how this happened as well as what types of
arrangements can be made with the phone company (CenturyLink in Texas).

Some details:
* PBX is located in Texas
* Phone carrier is CenturyLink
* FreePBX distro running asterisk 1.8.14
* source SIP extension is Mitel 5212, firmware 08.00.00.04, default
admin password (argh!). Phone is used by many different people.

More PBX setting details:
* inbound SIP traffic is not allowed through the firewall
* internal network is not accessed by many
* FreePBX web interface

*Questions I have at this moment:*
1) how were the calls placed? Was the Mitel SIP phone hacked somehow?
Asterisk PBX?


Check your logs. In the full log with verbosity 3 you can follow how 
calls were treated. Also the CDR should give you informations like the 
extension(s) who placed those calls


[...]

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Re: [asterisk-users] Disable fax detect on specific incoming DID

[Administrator TOOTAI]

Hi Noah,

Le 16/01/2015 23:13, Noah Engelberth a écrit :

The easiest way is to just run the Dial() command to forward the call to the 
hard fax without ever Answer()-ing the call.  Without an Answer() on the call, 
Asterisk can't listen for fax detection (because the call hasn't been set up 
and there is no audio leg yet).


It works perfectly :-), many thanks for the tip.

Regards

--
Daniel

[...]


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator 
TOOTAI
Sent: Friday, January 16, 2015 5:59 AM
To: Asterisk Users
Subject: [asterisk-users] Disable fax detect on specific incoming DID

Hello,

our gateway receive incoming calls from an outside gateway for multiple DIDs. 
For some of them we want fax detection, for other no. To do so, faxdetect is 
set to yes, but how to disable the fax detection for a specific incoming DID? 
For those DIDs, we just want to forward the call to a real fax machine DID 
which will do the job.

Thanks for any hint


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[asterisk-users] Disable fax detect on specific incoming DID

[Administrator TOOTAI]

Hello,

our gateway receive incoming calls from an outside gateway for multiple 
DIDs. For some of them we want fax detection, for other no. To do so, 
faxdetect is set to yes, but how to disable the fax detection for a 
specific incoming DID? For those DIDs, we just want to forward the call 
to a real fax machine DID which will do the job.


Thanks for any hint

Regards

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[asterisk-users] Asterisk 11.13 - No verbose logs

[Administrator TOOTAI]

Hi all,

On an Elastix server with asterisk 11.13.0 I have no verbose logs 
despite the fact that it's OK in CLI, eg verbose set to 3 in my case


Logger.conf

[logfiles]
;
; Format is "filename" and then "levels" of debugging to be included:
;debug
;notice
;warning
;error
;verbose
;
; Special filename "console" represents the system console
;
;debug => debug
; The DTMF log is very handy if you have issues with IVR's
;dtmf => dtmf
;console => notice,warning,error
;console => notice,warning,error,debug\
;messages => notice,warning,error
full => notice,warning,error,debug,verbose
verbose => verbose

Files verbose and full in /var/lib/asterisk contains only:

[Dec  3 12:47:55] Asterisk 11.13.0 built by palosanto @ 
rpmbuild32-2.elastix.palosanto.com on a i686 running Linux on 2014-10-02 
17:41:51 UTC


Files are owned by user/group asterisk, rw r mode. I tried to rotate 
logs, no changes. Disk is not full, fuul log file contains others infos.


Any clue?

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Re: [asterisk-users] PBX hacked: why hundred of calls to the same number ?

[Administrator TOOTAI]

Le 01/10/2014 11:40, Olivier a écrit :

Hi,


Hi



Someone reported me that from a PBX on which someone gained fraudulent
access, he could observe hundreds of calls to the same destination
number.

For curiosity's sake, I'm wondering why would this happen (dialing the
same number over and over) ?

Some special numbers generate here and there revenues for callees (and
not for callers).
Beside sharing interests with the callee that get those revenues, why
a hacker would like to dial the same numbers over and over ?


callee is also the bad men. Go and buy an 899 number in France, hack 
PBXS and call your number :-)


[...]

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Daniel

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Re: [asterisk-users] NOT able to call on local extensions while successfully call on external mobile no.(using SONETEL account)

[Administrator TOOTAI]

Le 13/09/2014 20:04, Alok Srivastava a écrit :

*Dear List*
Plz help, i am not much experienced with asterisk. i configured it on
ubuntu 12.04. no problem when i call any mobile no(0091XX) but
when i call on my local asterisk  no.(101,102 or 105) it is not
connecting giving error
"Auto fallthrough, channel 'SIP/lucknow-006f' status is 'CHANUNAVAIL'
*while when i call 200 it is playing audiofile successfully. Please help
*here is my sip.conf and extensions.conf.


Check with 'sip show peers' in asterisk CLI, your extensions are not 
registred (bombay,lucknow and test5)


--
Daniel

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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Administrator TOOTAI]

Le 02/09/2014 20:18, Khalid Touati a écrit :

so it seems Asterisk Versions does not support video I guess


Asterisk supports video. I'm using it with asterisk 1.4 1.8 and 11 with 
GrandStream phones (H263, H263+ and H264). Works perfectly





On Mon, Sep 1, 2014 at 9:30 AM, Khalid Touati mailto:khalidtou...@gmail.com>> wrote:

Any article that goes through this (seems to be tedious) task to add
video support and patents?


On Mon, Sep 1, 2014 at 8:14 AM, Joshua Colp mailto:jc...@digium.com>> wrote:

Khalid Touati wrote:

Hi Guys,


Kia ora,


Do you know of any asterisk community version that does
video codec
trans-coding or in other words supports video? I have
1.8.8.1 and I see
h263.c format files but can't see that codec in make
menuselect. it
might be just a license issue (if h263 has to have license),
but not
sure if community versions offer video calls at all.


Video transcoding is both usually patent encumbered as well as
computationally expensive. Asterisk supports passing through the
video untouched, but that's about it.

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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--
Khalid Touati




--
Khalid Touati





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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Administrator TOOTAI]

Le 02/09/2014 09:38, Nick Awesome a écrit :

So there is no way to do that with pjsip?


Sorry, I didn't read carefully the subject. I can't answer for pjsip. My 
bad :-(




On 02 Sep 2014, at 11:35, Administrator TOOTAI  wrote:


Le 02/09/2014 08:47, Nick Awesome a écrit :

Hello guys.


Hi



Have 2 external numbers that required registration on provider server,

trunk1: 734322600*05*@80.75.132.66
trunk2: 734322600*50*@80.75.132.66

Thing is I can’t figure out how to route them to different IVRs

by default Asterisk can’t match endpoint

Request from '' failed for
'80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
matching endpoint found

Can’t set /identify /by IP because they got the same ip.

Is there way to configure asterisk so incoming calls from same IP but
different ID will use different contexts?


You have to register to the gateway with each account user and password like

sip.conf

register = 734322600*05*:password1@myProvider/734322600*05*
register = 734322600*50*:password2@myProvider/734322600*50*

[myProvider]
type=peer
host=80.75.132.66
context=from-myProvider
...

extensions.conf

[from-myProvider]
exten = 734322600*05*,1,NoOp(Incoming call to 734322600*05*)
...

exten = 734322600*50*,1,NoOp(Incoming call to 734322600*50*)
...

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Re: [asterisk-users] PJSIP issues with handling incoming calls

[Administrator TOOTAI]

Le 02/09/2014 08:47, Nick Awesome a écrit :

Hello guys.


Hi



Have 2 external numbers that required registration on provider server,

trunk1: 734322600*05*@80.75.132.66
trunk2: 734322600*50*@80.75.132.66

Thing is I can’t figure out how to route them to different IVRs

by default Asterisk can’t match endpoint

Request from '' failed for
'80.75.132.66:5060' (callid: 50e9132765782741404408k2469rmwp) - No
matching endpoint found

Can’t set /identify /by IP because they got the same ip.

Is there way to configure asterisk so incoming calls from same IP but
different ID will use different contexts?


You have to register to the gateway with each account user and password like

sip.conf

register = 734322600*05*:password1@myProvider/734322600*05*
register = 734322600*50*:password2@myProvider/734322600*50*

[myProvider]
type=peer
host=80.75.132.66
context=from-myProvider
...

extensions.conf

[from-myProvider]
exten = 734322600*05*,1,NoOp(Incoming call to 734322600*05*)
...

exten = 734322600*50*,1,NoOp(Incoming call to 734322600*50*)
...

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[asterisk-users] MeetMe - Howto put in talk only mode using CLI/AMI

[Administrator TOOTAI]

Hi,

is there a way to put a conference participant in talk only mode (not 
listening) using CLI or AMI like mute/unmute ?


MeetMe in Asterisk 1.8

Thanks for any hint.

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