Re: [asterisk-users] chan_sip and 2 devices under same extension - transferring call endpoint(s)
Sounds like a job for TAPI. Google TAPI for Asterisk or Asterisk TSP I've been playing with SIPTAPI and it works pretty well. It's very simple to install and set up. Hope this Helps PC... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner Sent: Monday, December 29, 2014 9:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] chan_sip and 2 devices under same extension - transferring call endpoint(s) On Mon, Dec 29, 2014 at 7:26 AM, Lukasz Sokol el.es...@gmail.com mailto:el.es...@gmail.com wrote: As the handsets have no LCD's to show the dialled number, I want to give the workforce the ability to dial OUT using the softphone, (as in, copy/paste the number from the CRM software into softphone then *immediately* transfer the originated call 'endpoint' to the handset of the same 'user' extension, somehow, the question is, HOW ? We use FreePBX and a custom CRM. What we do is use the Asterisk Manager interface to create a call using the originate command. Asterisk dials the users handset, once they answer Asterisk then dials the outbound number. No need for any transferring. You could also look at Asterisk call files to originate the call. Ryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] status - Unmonitored, how to change it
Put qualify=yes in the peer definition in sip.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Tuesday, December 30, 2014 1:59 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] status - Unmonitored, how to change it How to change status of peers Unmonitored to monitored? Home users showing Unmonitored some display timing. Name/UsernameHost Mask Port Status zoiper_kathy/zo 112.200.83.69 (D) 255.255.255.255 9330 Unmonitored clinic_server(null) (D) 255.255.255.255 0 Unmonitored voip 184.89.249.114 (S) 255.255.255.255 4569 OK (91 ms) -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hangup Chanel when a peer unregisters
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades Sent: Wednesday, November 05, 2014 4:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hangup Chanel when a peer unregisters On 04/11/14 15:11, Pat Collins wrote: Hello group and thank you for the attention. I'm using Asterisk 11.12 running on Ubuntu Server 12.04 We have an issue with channels remaining open after a SIP peer unregisters. It seems that if the peer goes away before manually hanging up a call, the channel remains open until a hangup request is sent from the CLI. Is there any way to drop a channel when the peer using it disappears? Googled every phrase I could think of. No luck. Thank you! Pat Collins rtptimeout= in sip.conf will hangup a channel if no rtp is received for a period of time. Thanks for the response Gareth. The problem is that I may have a conference call up for days at a time. During this time, there may be no activity for hours. If the endpoint the endpoint is able to send RTP keepalive packets, your solution is spot on. Will have a look at it. Thanks again! PC... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup Chanel when a peer unregisters
Hello group and thank you for the attention. I'm using Asterisk 11.12 running on Ubuntu Server 12.04 We have an issue with channels remaining open after a SIP peer unregisters. It seems that if the peer goes away before manually hanging up a call, the channel remains open until a hangup request is sent from the CLI. Is there any way to drop a channel when the peer using it disappears? Googled every phrase I could think of. No luck. Thank you! Pat Collins -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] incoming calls fall into echo test mode
Perhaps assigned as a test number somewhere along the line? Are these ISDN, SIP, IAX calls? There are MANY smart people on this list. Maybe sharing the relevant configs and traces is a good place to start??? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norman Molhant Sent: Saturday, July 19, 2014 10:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] incoming calls fall into echo test mode Hello all, Weird trouble here: we have 60-some happy subscribers on a FreePBX box, each with its own phone number, with no problem at all, except for one (and only one) subscriber who has this problem: his outgoing calls are ok, but when someone dials his phone number (be it from our network or from any other place in the world), the caller ears the standard message signalling he has entered the echo test mode and must dial # to exit that mode. Most callers don't understand what's going on, then give up and hang up without dialling #. Very few dial # one or more times, then those few get our customer's phone ringing and are then able to reach our customer. I went through all the docs, wikis and discussions I found on the web, without finding any data on how to solve that problem. I tried many things on our FreePBX box and found out the problem seems somehow linked with the customer's extension (or phone number), not his inbound route (changing the latter has no effect on the problem). Creating a new extension with another phone number would solve the problem (I tried it and it works), but this customer wants to keep his current phone number and when I tried deleting his extension then creating a new one with his current phone number, the new extension presented the same problem as the previous one... Anyone knows what could cause such a problem and/or how to solve it ? Thanks, Norman. ad...@csur.ca -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need some PHP/AMI guidance please
Hello all,
I've got some PHP code that opens an AMI socket and does a ConfBridgeList
for a specific bridge ().
This all works just fine but I need to filter the information displayed to
only CallerIDName so I can see a complete list of names of participants.
After days of googling and playing with it, I'm no closer than I was when I
started.
I'm not at all married to a table. A simple list of names is fine...
Any help is much appreciated!
Pertinent code:
?php
$ami = fsockopen(127.0.0.1, 5038, $errno, $errstr);
if (!$ami) {
echo ERROR: $errno - $errstrbr /\n;
} else {
fwrite($ami, Action: Login\r\nUsername: someuser\r\nSecret:
somesecret\r\nEvents: off\r\n\r\n);
fwrite($ami, Action: ConfbridgeList\r\nConference: \r\n\r\n);
sleep(1);
$record = fread($ami,1024);
$record = explode(\r\n, $record);
echo META HTTP-EQUIV=Refresh CONTENT=\20\;
echo table border=\1\ style='color: black;';
foreach($record as $value){
if(!strlen(stristr($value,'Asterisk'))0
!strlen(stristr($value,'Response'))0
!strlen(stristr($value,'Message'))0
!strlen(stristr($value,'Event'))0
strlen(strpos($value,' '))0)
php_table($value);;
}
echo /table;
fclose($ami);
}
function php_table($value){
$row1 = true;
$value = explode( , $value);
foreach($value as $field){
if($row1){
echo trtd.$field./td;
$row1 = false;
}
else{
echo td.$field./td/tr;
$row1 = true;
}
}
}
?
I think the explode is where I should be looking but I'm very new to PHP
Thank you!
Pat...
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Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Thursday, January 16, 2014 8:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable? On 1/16/2014 6:57 PM, Dan Austin wrote: Patrick Lists wrote: On 16-01-14 21:37, Gergely Kiss wrote: Dear List, I'm about to build an Asterisk 11.7 based PBX from scratch for our company. I'm in the middle of the planning phase and it turned out that our VoIP provider prefers H.323 protocol for handling voice calls (while SIP is also supported as plan B). It's SIP everywhere and anyone who requires you, in 2014, to use H.323 should get a clue. Avoid them or at least demand SIP Bah. There is nothing wrong with a working H.323 stack. Just assuming that they will have a working SIP stack because of the date can lead to heartache. As I never worked with H.323 channels in Asterisk earlier, I'm not sure if it's stable enough to be used in production. No idea. Maybe someone else with H.323 experience will respond. AFAIK it's a dead-end. The ooh323 channel has been fairly reliable in our use case, which involve connecting to a commercial IP PBX with crud SIP support. Only you can tell if it will work for you however, as sadly many times new core features only get tested against the SIP channel(s), or worse only implemented there as well. Our current Asterisk version is 11.5.1 Dan Sorry, have nothing to say of 11.5 but OOH323 works great in 1.8. I use it as an Avaya IP Office trunk. No problems. As you observed for yourself when you researched the topic there is not a lot of help available, and Asterisk team prefers to make everybody think that SIP is the only viable call setup protocol around. They kind of not talking a lot about their own IAX any more. The official H.323 is abandoned. OOH323 is being supported by a very capable and responsive guy. He does not frequent the user list as he subscribes to the developer list, so I normally transfer the help inquiries to him if there is no traction here. -Vladimir Hey Vladimir, can you share a bit about the ooH323 trunk to IPO configuration that's stable? I've tried a few different setups on both sides and wound up using a PRI to do it. A call from the IPO to Asterisk (1.6.2 at the time) would crash the Asterisk box or not work at all. I'd love to be able to offer this to my IPO and CM customers! Thank you! Pat... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cut off last character of EXTEN
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, August 20, 2013 4:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Cut off last character of EXTEN
Hello,
how can I cut off the last character of the EXTEN-variable with variating
length ?
So I have :
112233#
123#
123456789#
I want to cut off the last character.
${EXTEN:-1} gives me #, but that is the character I want to cut off.
Kind regards,
Jonas.
Here ya go:
112233# use ${EXTEN:0:6})
123# use ${EXTEN:0:3})
123456789# use ${EXTEN:0:9})
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Re: [asterisk-users] Cut off last character of EXTEN
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes
Sent: Tuesday, August 20, 2013 7:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cut off last character of EXTEN
On 20 Aug 2013, at 12:25, Pat Collins wrote:
Here ya go:
112233# use ${EXTEN:0:6})
123# use ${EXTEN:0:3})
123456789# use ${EXTEN:0:9})
I think 'variable length' implied 'unknown length'...
S
Yea, I realized that after I replied
Got ahead of myself again.
That was the only way I was able to get rid of the '#' tho
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Re: [asterisk-users] auto answer
Dialplan auto answer
; intercom
exten=_*,1,SIPAddHeader(Call-Info:
sip:xxx.xxx.xxx.xxx\;answer-after=0) ;xxx.xxx.xxx.xxx is the address of
your asterisk box
exten=_*,n,Dial(SIP/${EXTEN:1})
As long as your phones are compatible, this MIGHT work.
Worked for me. Sadly, I cannot recall which phones we were using. Long
time ago.
Hope it helps,
Pat
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan
N
Sent: Wednesday, July 17, 2013 9:02 AM
To: bilal ghayyad; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] auto answer
yes its not asterisk configuration, its phone feature and phone
configuration.
On Wed, Jul 17, 2013 at 3:27 PM, bilal ghayyad bilmar...@yahoo.com wrote:
So it is not at asterisk configuration?
Regards
Bilal
_
From: A J Stiles asterisk_l...@earthshod.co.uk
To: bilal ghayyad bilmar...@yahoo.com; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Wednesday, July 17, 2013 12:57 PM
Subject: Re: [asterisk-users] auto answer
On Wednesday 17 July 2013, bilal ghayyad wrote:
But this not in the sip.conf, this in the extensions.conf, right?
Regards
Bilal
No. This would be set up in the phone's own configuration file, which in
turn
depends on the make and model of phone (and its location depends on your
site
setup).
--
AJS
Answers come *after* questions.
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[asterisk-users] AMI help needed
Hello group, I put together a simple PHP based conferencing manager for Asterisk 11.3 I used ODBC MYSQL for conference IDs and PINs. All this is working as desired but I would love to add an active conferences display to the front end. It seems to me that AMI is the way to go but I have no idea how to accomplish this or even where to begin. Any guidance is appreciated. Pat... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ODBC dialplan looping problem
) in new stack //LAST GOOD RESULT!!! atpconf001*CLI -- Executing [444999@getpin:5] GotoIf(SIP/testbridge2-0021, 0?cleanup,1) in new stack -- Executing [444999@getpin:6] GotoIf(SIP/testbridge2-0021, 0?good_exten,1) in new stack -- Executing [444999@getpin:7] Goto(SIP/testbridge2-0021, loop_start) in new stack atpconf001*CLI -- Goto (getpin,444999,3) -- Executing [444999@getpin:3] NoOp(SIP/testbridge2-0021, ) in new stack atpconf001*CLI -- Executing [444999@getpin:4] Set(SIP/testbridge2-0021, ROW_RESULT=) in new stack //BAD RESULTS FOREVER!!! atpconf001*CLI -- Executing [444999@getpin:5] GotoIf(SIP/testbridge2-0021, 0?cleanup,1) in new stack atpconf001*CLI -- Executing [444999@getpin:6] GotoIf(SIP/testbridge2-0021, 0?good_exten,1) in new stack atpconf001*CLI -- Executing [444999@getpin:7] Goto(SIP/testbridge2-0021, loop_start) in new stack -- Goto (getpin,444999,3) -- Executing [444999@getpin:3] NoOp(SIP/testbridge2-0021, ) in new stack //AND SO ON Thank you!! Pat Collins... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC dialplan looping problem
Thank you Bharat.
Sadly, that made no difference.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharat
Lalcheta
Sent: Thursday, April 18, 2013 7:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ODBC dialplan looping problem
I think there is no problem with asterisk.
exten=_XX,n,GotoIf($[${ODBC_FETCH} = FAILURE]?cleanup,1)
exten=_XX,n,GotoIf($[${ROW_RESULT} = ${CONF_PIN}]?good_exten,1)
It should be,
exten=_XX,n,GotoIf($[${ODBC_FETCH} = FAILURE]?getpin,cleanup,1)
exten=_XX,n,GotoIf($[${ROW_RESULT} =
${CONF_PIN}]?getpin,good_exten,1)
Hope it helps,
Regards,
Bharat Lalcheta
On Thu, Apr 18, 2013 at 4:45 PM, Pat Collins drdialt...@optonline.net
wrote:
All,
Thank you in advance for any help.
I have a customer in need of a conferencing system. A requirement is for
users to each have their own PIN for the same bridge.
So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC
connector to parse the table.
Asterisk is connected and reads the rows as expected. The problem is that
if a user enters a PIN that is NOT in the table, asterisk goes crazy and
continues to loop forever.
Please have a look and tell me where I went so wrong.
Func_odbc.conf looks like this:
[PIN]
dsn=BRIDGE
mode=multirow
readsql=SELECT pin from users WHERE confid='${SQL_ESC(${CONF_ID})}'
extensions.conf section:
[infromhost] ;Host dials over SIP trunk exten=,1,Answer
exten=,n,Background(conf-getconfno)
exten=,n,WaitExten(10)
exten=,n,Hangup
exten=_XX,1,Set(GLOBAL(CONF_ID)=${EXTEN})
exten=_XX,n,GoTo(rooms,${EXTEN},1)
;
[rooms]
exten=_XX,1,Set(CONF_ID=${EXTEN})
exten=_XX,n,Background(conf-getpin)
exten=_XX,n,WaitExten(5)
exten=_XX,n,Hangup
exten=_1X,1,Goto(getpin,${EXTEN},1)
exten=_2X,1,Goto(getpin,${EXTEN},1)
exten=_3X,1,Goto(getpin,${EXTEN},1)
exten=_4X,1,Goto(getpin,${EXTEN},1)
exten=_5X,1,Goto(getpin,${EXTEN},1)
exten=_6X,1,Goto(getpin,${EXTEN},1)
exten=_7X,1,Goto(getpin,${EXTEN},1)
exten=_8X,1,Goto(getpin,${EXTEN},1)
exten=_9X,1,Goto(getpin,${EXTEN},1)
exten=i,1,Goto(getpin,${CONF_PIN},1)
;
[getpin]
exten=_XX,1,Set(GLOBAL(CONF_PIN)=${EXTEN})
exten=_XX,n,Set(ODBC_ID=${ODBC_PIN(1)})
exten=_XX,n(loop_start),NoOp()
exten=_XX,n,Set(ROW_RESULT=${ODBC_FETCH(${ODBC_ID})})
exten=_XX,n,GotoIf($[${ODBC_FETCH} = FAILURE]?cleanup,1)
exten=_XX,n,GotoIf($[${ROW_RESULT} = ${CONF_PIN}]?good_exten,1)
exten=_XX,n,Goto(loop_start)
;
exten=cleanup,1,Verbose(1,Finish up)
same=n,Verbose(1,PIN not found)
same=n,ODBCFinish(${ODBC_ID})
same=n,playback(conf-invalidpin)
same=n,Goto(rooms,${CONF_ID}1)
same=n,Hangup()
;
exten=good_exten,1,Verbose(1,The PIN is available)
same=n,ODBCFinish(${ODBC_ID})
same=n,Verbose(1,Drop Caller into the bridge)
same=n,Set(CONFBRIDGE(user,template)=default_user)
same=n,ConfBridge(${CONF_ID},default_bridge,,sample_user_menu)
same=n,Hangup()
;
The log shows the 3 existing DB table rows are found but continues to cycle
indefinitely if the PIN is NOT found.
First few rows of the console log:
=
Connected to Asterisk 11.3.0 currently running on atpconf001 (pid = 1695)
atpconf001*CLI
== Using SIP RTP CoS mark 5
atpconf001*CLI
-- Executing [067740@default:1] Set(SIP/testbridge2-0021,
GLOBAL(CONF_ID)=067740) in new stack
== Setting global variable 'CONF_ID' to '067740'
-- Executing [067740@default:2] Goto(SIP/testbridge2-0021,
rooms,067740,1) in new stack
atpconf001*CLI
-- Goto (rooms,067740,1)
-- Executing [067740@rooms:1] Set(SIP/testbridge2-0021,
CONF_ID=067740) in new stack
-- Executing [067740@rooms:2] BackGround(SIP/testbridge2-0021,
conf-getpin) in new stack
atpconf001*CLI
-- SIP/testbridge2-0021 Playing 'conf-getpin.slin' (language 'en')
atpconf001*CLI
-- Executing [067740@rooms:3] WaitExten(SIP/testbridge2-0021, 5)
in new stack
atpconf001*CLI
== CDR updated on SIP/testbridge2-0021
-- Executing [444999@rooms:1] Goto(SIP/testbridge2-0021,
getpin,444999,1) in new stack
-- Goto (getpin,444999,1)
-- Executing [444999@getpin:1] Set(SIP/testbridge2-0021,
GLOBAL(CONF_PIN)=444999) in new stack
== Setting global variable 'CONF_PIN' to '444999'
atpconf001*CLI
-- Executing [444999@getpin:2] Set(SIP/testbridge2-0021,
ODBC_ID=32) in new stack
-- Executing [444999@getpin:3] NoOp(SIP/testbridge2-0021, ) in
new stack
-- Executing [444999@getpin:4] Set(SIP/testbridge2-0021,
ROW_RESULT=112233) in new stack
atpconf001*CLI
-- Executing [444999@getpin:5] GotoIf(SIP/testbridge2-0021,
0?cleanup,1) in new stack
-- Executing [444999@getpin:6] GotoIf(SIP/testbridge2-0021,
0?good_exten
Re: [asterisk-users] ODBC dialplan looping problem
Thanks for the reply Doug!
How might I incorporate this into my dialplan?
Sorry, learning as I go
On a side note, any idea how to strip off the # at the end of a string?
${EXTEN:0:6} doesn't do the trick.
Is this even possible?
Pat...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Thursday, April 18, 2013 7:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ODBC dialplan looping problem
The problem is that if a user enters a PIN that is NOT in the table,
asterisk goes crazy and continues to loop forever.
Why don't you use read instead? This is what I have:
exten = s,n,Answer(500)
exten = s,n,Read(get-room-num,conf-getconfno)
; *
; Get conference room number, if number entered is 9 **
; jump to verify.**
; *
exten = s,n,NoOP(${conf-getchannel})
Do some mysql magic here
exten = s,n,GotoIf($[${conference.room} != ]?s-process,1:s-notexist,1)
Doug
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Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] ODBC dialplan looping problem
Perfect
THANK YOU!!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharat
Lalcheta
Sent: Thursday, April 18, 2013 8:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ODBC dialplan looping problem
Sorry for that.
I believe that, dialplan misbehave only when you didn't get the result
right?...then instead of below
exten=_XX,n,GotoIf($[${ODBC_FETCH} = FAILURE]?cleanup,1)
try this...
exten=_XX,n,GotoIf($[${ODBC_FETCH_STATUS} = FAILURE]?cleanup,1)
Hope it helps you out.
Regards,
Bharat Lalcheta
On Thu, Apr 18, 2013 at 5:36 PM, Pat Collins drdialt...@optonline.net
wrote:
Thank you Bharat.
Sadly, that made no difference.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharat
Lalcheta
Sent: Thursday, April 18, 2013 7:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] ODBC dialplan looping problem
I think there is no problem with asterisk.
exten=_XX,n,GotoIf($[${ODBC_FETCH} = FAILURE]?cleanup,1)
exten=_XX,n,GotoIf($[${ROW_RESULT} = ${CONF_PIN}]?good_exten,1)
It should be,
exten=_XX,n,GotoIf($[${ODBC_FETCH} = FAILURE]?getpin,cleanup,1)
exten=_XX,n,GotoIf($[${ROW_RESULT} =
${CONF_PIN}]?getpin,good_exten,1)
Hope it helps,
Regards,
Bharat Lalcheta
On Thu, Apr 18, 2013 at 4:45 PM, Pat Collins drdialt...@optonline.net
wrote:
All,
Thank you in advance for any help.
I have a customer in need of a conferencing system. A requirement is for
users to each have their own PIN for the same bridge.
So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC
connector to parse the table.
Asterisk is connected and reads the rows as expected. The problem is that
if a user enters a PIN that is NOT in the table, asterisk goes crazy and
continues to loop forever.
Please have a look and tell me where I went so wrong.
Func_odbc.conf looks like this:
[PIN]
dsn=BRIDGE
mode=multirow
readsql=SELECT pin from users WHERE confid='${SQL_ESC(${CONF_ID})}'
extensions.conf section:
[infromhost] ;Host dials over SIP trunk exten=,1,Answer
exten=,n,Background(conf-getconfno)
exten=,n,WaitExten(10)
exten=,n,Hangup
exten=_XX,1,Set(GLOBAL(CONF_ID)=${EXTEN})
exten=_XX,n,GoTo(rooms,${EXTEN},1)
;
[rooms]
exten=_XX,1,Set(CONF_ID=${EXTEN})
exten=_XX,n,Background(conf-getpin)
exten=_XX,n,WaitExten(5)
exten=_XX,n,Hangup
exten=_1X,1,Goto(getpin,${EXTEN},1)
exten=_2X,1,Goto(getpin,${EXTEN},1)
exten=_3X,1,Goto(getpin,${EXTEN},1)
exten=_4X,1,Goto(getpin,${EXTEN},1)
exten=_5X,1,Goto(getpin,${EXTEN},1)
exten=_6X,1,Goto(getpin,${EXTEN},1)
exten=_7X,1,Goto(getpin,${EXTEN},1)
exten=_8X,1,Goto(getpin,${EXTEN},1)
exten=_9X,1,Goto(getpin,${EXTEN},1)
exten=i,1,Goto(getpin,${CONF_PIN},1)
;
[getpin]
exten=_XX,1,Set(GLOBAL(CONF_PIN)=${EXTEN})
exten=_XX,n,Set(ODBC_ID=${ODBC_PIN(1)})
exten=_XX,n(loop_start),NoOp()
exten=_XX,n,Set(ROW_RESULT=${ODBC_FETCH(${ODBC_ID})})
exten=_XX,n,GotoIf($[${ODBC_FETCH} = FAILURE]?cleanup,1)
exten=_XX,n,GotoIf($[${ROW_RESULT} = ${CONF_PIN}]?good_exten,1)
exten=_XX,n,Goto(loop_start)
;
exten=cleanup,1,Verbose(1,Finish up)
same=n,Verbose(1,PIN not found)
same=n,ODBCFinish(${ODBC_ID})
same=n,playback(conf-invalidpin)
same=n,Goto(rooms,${CONF_ID}1)
same=n,Hangup()
;
exten=good_exten,1,Verbose(1,The PIN is available)
same=n,ODBCFinish(${ODBC_ID})
same=n,Verbose(1,Drop Caller into the bridge)
same=n,Set(CONFBRIDGE(user,template)=default_user)
same=n,ConfBridge(${CONF_ID},default_bridge,,sample_user_menu)
same=n,Hangup()
;
The log shows the 3 existing DB table rows are found but continues to cycle
indefinitely if the PIN is NOT found.
First few rows of the console log:
=
Connected to Asterisk 11.3.0 currently running on atpconf001 (pid = 1695)
atpconf001*CLI
== Using SIP RTP CoS mark 5
atpconf001*CLI
-- Executing [067740@default:1] Set(SIP/testbridge2-0021,
GLOBAL(CONF_ID)=067740) in new stack
== Setting global variable 'CONF_ID' to '067740'
-- Executing [067740@default:2] Goto(SIP/testbridge2-0021,
rooms,067740,1) in new stack
atpconf001*CLI
-- Goto (rooms,067740,1)
-- Executing [067740@rooms:1] Set(SIP/testbridge2-0021,
CONF_ID=067740) in new stack
-- Executing [067740@rooms:2] BackGround(SIP/testbridge2-0021,
conf-getpin) in new stack
atpconf001*CLI
-- SIP/testbridge2-0021 Playing 'conf-getpin.slin' (language 'en')
atpconf001*CLI
-- Executing [067740@rooms:3] WaitExten(SIP/testbridge2-0021, 5)
in new stack
atpconf001*CLI
== CDR updated on SIP/testbridge2-0021
-- Executing [444999
Re: [asterisk-users] Need advice on how to implement this ...
Have you looked into SLA? I have had good results with it. Will let asterisk act like a key system. Sent from Samsung tablet Chris Gentle gent...@gmail.com wrote: I need some advice on how to implement something in my dialplan. Here's the scenario. A call comes in on my [incoming] context and I answer it. The call turns out to be for my wife and she needs to answer it on a different handset somewhere else in the house. I've tried call parking but the wife acceptance factor is kind of low because we don't do it often enough for her to remember how to park and unpark. What I'd really like to do is define an easy DTMF sequence in features.conf (like 00) that would send the call back into my [incoming] context again, just like it was a new incoming call. Then it could be picked up anywhere in the house. What's the best way to go about this? I tried doing an AGI script that sets context/extension/priority to where I'd like for it to go but it doesn't seem to work. Am I on the right track or is there a better way to do this? -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending calls from behind NAT
Check reinvite and NAT settings on the line as well as the SIP peers. You can use a stun client from inside your network to see what’s going on with the NAT From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Budinick Sent: Tuesday, November 13, 2012 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sending calls from behind NAT I'm with Duncan, you need a public IP address, not private. Chris Budinick Network Technician RAINIER CONNECT _ From: Duncan Turnbull dun...@e-simple.co.nz To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, November 13, 2012 1:29:28 PM Subject: Re: [asterisk-users] Sending calls from behind NAT On 14/11/2012, at 10:16 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the difference. Really, I did not understand what exactly they need. But maybe what they need is to see my public IP address without the private IP address (this what I understood if I am right). I tried to use the following in the [general] settings in the sip.conf localnet=192.168.10.2/255.255.255.254 externadd =196.40.164.239 This should be externip not externadd You are still sending them your local address Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users attachment: image001.jpg-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] username ignored when trying to auth incoming invites
Try defaultuser=test instead of username=test -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Wolthuis Sent: Thursday, October 04, 2012 11:01 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] username ignored when trying to auth incoming invites Hello All, I am trying to debug an odd issue. I have two UACs that are sending INVITEs to my asterisk 1.8 server. I want to start authenticating these incoming invite requests with digest auth. The UACs are not registered and I am using host ip to match them with a sip.conf peer. The issue I am seeing is that an incoming invite matches a specific peer (by host ip), but refuses to use the username parameter value for digest auth, it will only use the peer name. I see the following error: chan_sip.c: username mismatch, have node-a, digest has test I have the following sip.conf: [node-a] type=friend disallow=all allow=ulaw context=incoming-context host=XXX.XXX.XXX.XXX transport=udp username=test secret=1234 [node-b] type=friend disallow=all allow=ulaw context=incoming-context host=YYY.YYY.YYY.YYY transport=udp username=test secret=1234 If I auth using node-a as the username when sending an invite from that host, everything works. If I auth with test as the username from node-a, it fails with the error above. It appears that peer name is always being used for digest auth, rather than the contents of username. Is username the wrong place to specify this? Thanks for your help! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI help please
Thank you!
So, this code in dahdi-base.c should work?
void dahdi_rbsbits(struct dahdi_chan *chan, int cursig)
{
unsigned long flags;
if (cursig == chan-rxsig)
return;
if ((chan-flags DAHDI_FLAG_SIGFREEZE)) return;
spin_lock_irqsave(chan-lock, flags);
switch(chan-sig) {
case DAHDI_SIG_EM:
if (!(cursig DAHDI_XBIT)) {
__dahdi_hooksig_pvt(chan, DAHDI_RXSIG_START);
break;
}
/* Fall through */
Please forgive my ignorance.
As long as folks like you are there to help, folks like me will not remain
ignorant for long!
Thank you again!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell
Sent: Wednesday, October 03, 2012 1:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DAHDI help please
On Tue, Oct 02, 2012 at 11:22:31PM -0400, Pat Collins wrote:
Shaun,
To make more sense of the code, I changed
#define DAHDI_XBIT(3 2) to
#define DAHDI_XBIT(0)
Sadly, incoming calls do not work. Not sure exactly how to START or
RING when the RX AB bits are 00 Any ideas?
Thanks again for your help!
The board drivers call dahdi_rbsbits() when they want to report a change in
the state of the RBS bits for a channel. If you look in the code there you
will see where events are generated depending on the signalling type.
I should have pointed out that function in my previous email.
Cheers,
Shaun
--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
www.digium.com www.asterisk.org
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Re: [asterisk-users] DAHDI help please
Thank you for the reply!
So far, I've managed to get the on off hook to work properly!
My next problem is the incoming ring. I've changed the /include/dahdi/user.h
file:
#define DAHDI_ABIT (1 3)
#define DAHDI_BBIT (1 2)
#define DAHDI_CBIT (1 1)
#define DAHDI_DBIT (1 0)
#define DAHDI_XBIT (3 2) ADDED THIS!!
#define DAHDI_BITS_ABCD (DAHDI_ABIT | DAHDI_BBIT | DAHDI_CBIT | DAHDI_DBIT)
#define DAHDI_BITS_ABD (DAHDI_ABIT | DAHDI_BBIT | DAHDI_DBIT)
#define DAHDI_BITS_ACD (DAHDI_ABIT | DAHDI_CBIT | DAHDI_DBIT)
#define DAHDI_BITS_BCD (DAHDI_BBIT | DAHDI_CBIT | DAHDI_DBIT)
#define DAHDI_BITS_AC (DAHDI_ABIT | DAHDI_CBIT)
#define DAHDI_BITS_BD (DAHDI_BBIT | DAHDI_DBIT)
#define DAHDI_BITS_X (DAHDI_XBIT) AND THIS!!
Then in /driver/dahdi/dahdi-base.c I changed:
static void dahdi_rbs_sethook(struct dahdi_chan *chan, int txsig, int
txstate,
int timeout)
{
static const struct {
unsigned int sig_type;
/* Index is dahdi_txsig enum */
unsigned int bits[DAHDI_TXSIG_TOTAL];
} outs[NUM_SIGS] = {
{
/*
* We set the idle case of the DAHDI_SIG_NONE to
this pattern to make idle E1 CAS
* channels happy. Should not matter with T1, since
on an un-configured channel,
* who cares what the sig bits are as long as they
are stable
*/
.sig_type = DAHDI_SIG_NONE,
.bits[DAHDI_TXSIG_ONHOOK] = DAHDI_BITS_ABCD,
}, {
.sig_type = DAHDI_SIG_EM,
.bits[DAHDI_TXSIG_OFFHOOK] = DAHDI_SIG_NONE, /*
changed from ACD */
.bits[DAHDI_TXSIG_START] = DAHDI_SIG_NONE, /*
changed from ACD */
.bits[DAHDI_TXSIG_ONHOOK] = DAHDI_BITS_ABCD, /*
added this !! */
.bits[DAHDI_RXSIG_START]= DAHDI_BITS_X, /*
start.. xbit */
.bits[DAHDI_RXSIG_RING]= DAHDI_BITS_X, /*
ring.. xbit */
}, {
And this:
void dahdi_rbsbits(struct dahdi_chan *chan, int cursig)
{
unsigned long flags;
if (cursig == chan-rxsig)
return;
if ((chan-flags DAHDI_FLAG_SIGFREEZE)) return;
spin_lock_irqsave(chan-lock, flags);
switch(chan-sig) {
case DAHDI_SIG_EM: /* was FXO Groundstart */
/* B-bit only matters for FXO GS changed BBIT to
XBIT */
if (!(cursig DAHDI_XBIT)) {
__dahdi_hooksig_pvt(chan, DAHDI_RXSIG_START);
break;
}
/* Fall through */
case DAHDI_SIG_FXOGS: /* changed from E and M */
case DAHDI_SIG_EM_E1:
case DAHDI_SIG_FXOLS: /* FXO Loopstart */
case DAHDI_SIG_FXOKS: /* FXO Kewlstart */
if (cursig DAHDI_ABIT) /* off hook */
__dahdi_hooksig_pvt(chan,DAHDI_RXSIG_OFFHOOK);
else /* on hook */
__dahdi_hooksig_pvt(chan,DAHDI_RXSIG_ONHOOK);
break;
Thank you for your help Shaun!!!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell
Sent: Tuesday, October 02, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DAHDI help please
On Mon, Oct 01, 2012 at 06:44:22PM -0400, Pat Collins wrote:
Can anyone tell me if it is possible to invert the signaling bits on a
T1 channel?
I need to emulate PLAR signaling in asterisk. EM seems to be an
exact match if reversed.
I need idle bits and seized
Perhaps you could edit dahdi_rbs_sethook() and dahdi_q_sig()? Those
function map state to RBS states.
--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
www.digium.com www.asterisk.org
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Re: [asterisk-users] DAHDI help please
Shaun,
To make more sense of the code, I changed
#define DAHDI_XBIT (3 2) to
#define DAHDI_XBIT (0)
Sadly, incoming calls do not work. Not sure exactly how to START or RING
when the RX AB bits are 00
Any ideas?
Thanks again for your help!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pat Collins
Sent: Tuesday, October 02, 2012 2:48 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DAHDI help please
Thank you for the reply!
So far, I've managed to get the on off hook to work properly!
My next problem is the incoming ring. I've changed the /include/dahdi/user.h
file:
#define DAHDI_ABIT (1 3)
#define DAHDI_BBIT (1 2)
#define DAHDI_CBIT (1 1)
#define DAHDI_DBIT (1 0)
#define DAHDI_XBIT (3 2) ADDED THIS!!
#define DAHDI_BITS_ABCD (DAHDI_ABIT | DAHDI_BBIT | DAHDI_CBIT | DAHDI_DBIT)
#define DAHDI_BITS_ABD (DAHDI_ABIT | DAHDI_BBIT | DAHDI_DBIT) #define
DAHDI_BITS_ACD (DAHDI_ABIT | DAHDI_CBIT | DAHDI_DBIT) #define DAHDI_BITS_BCD
(DAHDI_BBIT | DAHDI_CBIT | DAHDI_DBIT) #define DAHDI_BITS_AC (DAHDI_ABIT |
DAHDI_CBIT) #define DAHDI_BITS_BD (DAHDI_BBIT | DAHDI_DBIT)
#define DAHDI_BITS_X (DAHDI_XBIT) AND THIS!!
Then in /driver/dahdi/dahdi-base.c I changed:
static void dahdi_rbs_sethook(struct dahdi_chan *chan, int txsig, int
txstate,
int timeout)
{
static const struct {
unsigned int sig_type;
/* Index is dahdi_txsig enum */
unsigned int bits[DAHDI_TXSIG_TOTAL];
} outs[NUM_SIGS] = {
{
/*
* We set the idle case of the DAHDI_SIG_NONE to
this pattern to make idle E1 CAS
* channels happy. Should not matter with T1, since
on an un-configured channel,
* who cares what the sig bits are as long as they
are stable
*/
.sig_type = DAHDI_SIG_NONE,
.bits[DAHDI_TXSIG_ONHOOK] = DAHDI_BITS_ABCD,
}, {
.sig_type = DAHDI_SIG_EM,
.bits[DAHDI_TXSIG_OFFHOOK] = DAHDI_SIG_NONE, /*
changed from ACD */
.bits[DAHDI_TXSIG_START] = DAHDI_SIG_NONE, /*
changed from ACD */
.bits[DAHDI_TXSIG_ONHOOK] = DAHDI_BITS_ABCD, /*
added this !! */
.bits[DAHDI_RXSIG_START]= DAHDI_BITS_X, /*
start.. xbit */
.bits[DAHDI_RXSIG_RING]= DAHDI_BITS_X, /*
ring.. xbit */
}, {
And this:
void dahdi_rbsbits(struct dahdi_chan *chan, int cursig) {
unsigned long flags;
if (cursig == chan-rxsig)
return;
if ((chan-flags DAHDI_FLAG_SIGFREEZE)) return;
spin_lock_irqsave(chan-lock, flags);
switch(chan-sig) {
case DAHDI_SIG_EM: /* was FXO Groundstart */
/* B-bit only matters for FXO GS changed BBIT to
XBIT */
if (!(cursig DAHDI_XBIT)) {
__dahdi_hooksig_pvt(chan, DAHDI_RXSIG_START);
break;
}
/* Fall through */
case DAHDI_SIG_FXOGS: /* changed from E and M */
case DAHDI_SIG_EM_E1:
case DAHDI_SIG_FXOLS: /* FXO Loopstart */
case DAHDI_SIG_FXOKS: /* FXO Kewlstart */
if (cursig DAHDI_ABIT) /* off hook */
__dahdi_hooksig_pvt(chan,DAHDI_RXSIG_OFFHOOK);
else /* on hook */
__dahdi_hooksig_pvt(chan,DAHDI_RXSIG_ONHOOK);
break;
Thank you for your help Shaun!!!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell
Sent: Tuesday, October 02, 2012 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DAHDI help please
On Mon, Oct 01, 2012 at 06:44:22PM -0400, Pat Collins wrote:
Can anyone tell me if it is possible to invert the signaling bits on a
T1 channel?
I need to emulate PLAR signaling in asterisk. EM seems to be an
exact match if reversed.
I need idle bits and seized
Perhaps you could edit dahdi_rbs_sethook() and dahdi_q_sig()? Those
function map state to RBS states.
--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
www.digium.com www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
[asterisk-users] DAHDI help please
Can anyone tell me if it is possible to invert the signaling bits on a T1 channel? I need to emulate PLAR signaling in asterisk. EM seems to be an exact match if reversed. I need idle bits and seized Thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
