Re: [asterisk-users] Feature Request: GotoIfTimeWithOffset
2009/12/18 Barry Miller asterisk-us...@notanet.net
On Fri, Dec 18, 2009 at 12:23:22AM +0100, Olivier wrote:
Today, this IVR is using function AEL GotoIfTime in several places.
The problem is if it's 11pm at the moment I'm testing this IVR, I can't
nicely test the 9am or 2pm branch.
Suggestions ?
How about setting, say, LUNCHTIME to 23:00-23:59 and using
GotoIfTime(${LUNCHTIME},...) until you're ready to go live?
I try to do this when there are multiple GotoIfTime's referencing
the same interval in a dialplan. Companies change their minds about
things like lunch times, working hours, shift changes all the time,
and this makes it easier to change down the road.
You're right.
In my case I've got a dozen of opening and closing times :
9h00 - 12h00 for service1
9h00 - 13h00 for service2
...
When 2 services MUST share the same opening and closing times, then your
suggestion is very interesting.
When 2 services HAPPEN to share the same opening and closing times, IMHO,
keeping separate variables seems more adapted.
What makes me express this Feature Request is that, for testing purpose,
changing 10 time variables without any mistake is possible but difficult :
having a separate offset parameter you can change at will, seems more secure
and easy.
Chad's suggestion (focusing on an offset for hours) is very smart as it
relies on existing dialplan functions and serves my original goal (create
artificial conditions for testing without changing a line in dialplan after
testing).
I will also patch my Asterisk source code and report here, my experience to
Tilghman's patch.
Thanks for all
--Barry
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Re: [asterisk-users] Feature Request: GotoIfTimeWithOffset
On Thu, Dec 17, 2009 at 07:19:49PM -0600, Tilghman Lesher wrote: On Thursday 17 December 2009 17:23:22 Olivier wrote: When I was testing an IVR, I realized I miss a function I would call GotoIfTimeWithOffset. Today, this IVR is using function AEL GotoIfTime in several places. The problem is if it's 11pm at the moment I'm testing this IVR, I can't nicely test the 9am or 2pm branch. GotoIfTimeWithOffset would get 2 incoming arguments : - the first is a time range (just like GotoIfTime), - the second is a duration offset which you could delay or rewind time. After testing, you would just have to set this offset to 0, to get a production-ready dialplan, without changing a line. https://issues.asterisk.org/view.php?id=16464 I've made some important changes to your idea. The method for invoking this is setting a function to a particular date and time. The reason for this is to permit testing of a live system, without changing the actual dialplan. You could set up a test with a particular extension that sets this function before jumping to the incoming context, or you could use conditional evaluation, such as checking callerid for the administrator's cellphone number, in the incoming context to evaluate whether to set this function or not. Looks nice. Just one point: This is a global (or rather: channel-specific, but still common to all in thecontext of that channel) setting. Suppose I have ( and suppose Asterisk even accepts this syntax) : $(TESTTIME(...)} GotoIfTime(test1,...) GotoIfTime(test2,...) Both test1 and test2 would be evaluated with the same offset as set in the TESTTIME() above. Is that what you want? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To Asterisk AMI Gurus - Tacking issue with originate
On Thu, Dec 17, 2009 at 08:29:55PM -0500, Bruce Nik wrote:
Hello Everyone,
I am making a simple index.php file which will allow a web user to enter his
$phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged.
Following is the index.php and the contents of extensions_custom.conf. When
I submit the form nothing happens. I don't even see Manager Connected msg.
Your input will be much appreciated. I am thinking I have some syntax
problem.
Time to debug the PHP part, then. This looks like an apache/PHP
question...
/etc/asterisk/extensions_custom.conf
*[testphp]
exten = _X.,1,Answer()
exten = _X.,n,Dial(SIP/testTrunk/${EXTEN})
exten = _X.,n,Hangup()*
*
*
*
*
*
*
*
/var/www/html/clickncall/index.php
html
head
titleClicknCall/title
/head
bodybrbr
div align=center
?php
$sys_ip = 127.0.0.1;
$User_str = testphp;
$Secret_str = testphp;
if ($_POST['x']) {
$oSocket = fsockopen($sys_ip, 5038, $errnum, $errdesc) or die(Connection to
host failed);
fputs($oSocket, Action: login\r\n);
fputs($oSocket, Username: $User_str\r\n);
fputs($oSocket, Secret: $Secret_str\r\n\r\n);
fputs($oSocket, Events: off\r\n\r\n);
fputs($oSocket, Action: originate\r\n);
fputs($oSocket, Channel: SIP/testTrunk/$phoneNumb\r\n);
fputs($oSocket, Exten: $dialNumb\r\n);
And you get those variables from where?
Are they submitted in the form?
(And if so: if you don't remove strange characters such as '\r\n' from
them, the user can use them to submit complete manager requests).
fputs($oSocket, Context: testphp\r\n);
fputs($oSocket, Priority: 1\r\n\r\n);
fputs($oSocket, Timeout: 1\r\n);
fputs($oSocket, CallerId: $spoofNumb\r\n);
fputs($oSocket, Async: true\r\n);
fputs($oSocket, Action: Logoff\r\n\r\n);
fclose($oSocket);
print $_POST['x'];
} else {
print form method=\post\action=\$_SERVER[PHP_SELF]\;
print br;
print PHONE Number: input type=\text\name=\phoneNumb\;
print br;
print PARTY Number: input type=\text\name=\dialNumb\;
print br;
print FORGE Number: input type=\text\name=\spoofNumb\;
print br;
print input type=\Submit\ value=\ Dial
\;
print /form;
} ?
/div
/body
/html
*/etc/asterisk/manager_custom.conf*
[testphp]
secret = testphp
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
To slightly reduce the chance of abuse:
read=
write=call
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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[asterisk-users] wrapuptime?
Hi! Trying to understand how wrapuptime is working... I have written a small php script that let agents log in/out off a queue. That part is working as a clock but wrapuptime is not doing what I expect. Input Interiör - Queue Manager 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (5s holdtime, 94s talktime), W:0, C:8, A:1, SL:0.0% within 0s Members: SIP/0317998971 with penalty 2 (dynamic) (Not in use) has taken no calls yet SIP/0317998975 with penalty 2 (dynamic) (Not in use) has taken no calls yet SIP/0317998972 with penalty 1 (dynamic) (Not in use) has taken 8 calls (last was 2 secs ago) No Callers SIP/0317998972 did hang up 8 seconds ago, but if someone calls the queue at this moment, the call will start ringing on SIP/0317998972 again. I thought the wrapuptime should cause the call to go to one of the other agents. Did I miss anything in the configs or is it that we have different penalties or...? cat queues.conf [general] ; autofill=yes keepstats=yes ; ; [0317998989] retry=5 strategy=rrmemory timeout=20 wrapuptime=120 cat agents.conf [general] ; persistentagents=yes ; ; [agents] ; agent = 0317998971,1234,Stefan Andersson agent = 0317998972,1234,Kerem Tubluk agent = 0317998975,1234,Magnus Benngard agent = 0317998976,1234,Jimmy Beckman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feature Request: GotoIfTimeWithOffset
Steve Edwards asterisk@sedwards.com writes: Wouldn't a set time function be more usefull? I really like that idea. Enough that I could try to lobby internally for funding, if you know someone who is willing to do the work... /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF doubler when using READ()
Hi there, I have some problems when using READ() statement in the dialplan to collect DTMF digits. I'm using the following within my extensions.conf to receive 6 digits exten = 9070,n,Read(CONFNO,conf-getpin,6) So far it works! The user getting the announcement and asterisk waits for 6 digits. User entered 102030, but asterisk gets 102033!!! Executing [9...@incoming:10] Read(SIP/1.1.1.1-08a3dfd0, CONFNO|conf-getpin|6) in new stack Accepting a maximum of 6 digits. SIP/1.1.1.1-08a3dfd0 Playing 'conf-getpin' (language 'en_GB') User entered '102033' It's not a problem caused by the user. I can reproduce this behavior and get a digit/DTMF doubler somewhere between the first and the last digit. More users on system means higher risk of a doubler. If there is just one user on the system in a conference getting MusicOnHold the CPU utilization is about 25%! System is Debian lenny (stable) Kernel: 2.6.26 #1 SMP asterisk: 1.4.21.2 Intel Dual Xeneon @2,8 3GB RAM Some ideas? Cheers Joern ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] test request for new event Pickup when a call is picked up from an other phone
A new patch has been made for an extra Manager Event when a call-pickup has occurred. There are two possible situations 1) by using *8 2) by using *8123 (to pickup extension 123 when it is ringing) The manager event looks like: Event: Pickup Privilege: call,all Channel: SIP/ast163-000c UniqueID: astium-21-1261065321.12 TargetChannel: SIP/ast165-000b TargetUniqueID: astium-21-1261065314.11 the patch is described in: https://issues.asterisk.org/view.php?id=16431 Ideally, if you can get someone from the asterisk-dev and/or asterisk-users mailing lists to test this out and report back here, that would be fantastic! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX for Asterisk
Where do you get FFA? I have not seen this, what is the minimum version of Asterisk that you need? Sorry about the questions. Thank you and have a nice day, Anthony Francis -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj Chand Sent: Thursday, December 17, 2009 8:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FAX for Asterisk Just finished with the instructions from digium website/ net on how to compile FFA: After restart, modules did not get loaded so tried to load manually: [Dec 18 14:31:26] WARNING[11002]: loader.c:359 load_dynamic_module: Error loadin ile: No such file or directory [Dec 18 14:31:26] WARNING[11002]: loader.c:653 load_resource: Module 'res_fax.so Verified the files exist: astbh00*CLI module load res_f res_fax.so res_features.so res_fax_digium.so astbh00*CLI module load res_f Help! :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX/NEW delays
Hi, Could someone tell me where are the good places in chan_iax to put trace points when I experience strange delays in NEW processing? I tried to output some debug after every stage of socket_process / case IAX_COMMAND_NEW, but it all takes max 30ms. However, sometimes in a normal call I get an extreme delay though (around 10 seconds), if I can believe the timestamp field... Do you know any good places to look at in ast-1.4? Or maybe someone had this problem before and can tell me what can cause delays that long. Thanks, Stan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To Asterisk AMI Gurus - Tacking issue with originate
Obvious debugging steps at this point:-
1. Manually connecting to AMI and testing the commands
1. telnet 127.0.0.1 5038
2. Login command
3. Originate command
2. Use a dummy AMI listener instead of actual asterisk to see if the
events are going through
1. Stop asterisk (or change the manager port)
2. nc -l -k 127.0.0.1 5038
3. Confirm that 'nc' has taken the tcp port 5038
netstat -nlp | grep 5038
4. Test the php application to see if the login request can be seen on
the console from where 'nc' was started
3. Edit /etc/php.ini (or similar file) to enable logging and then check
the log file :)
On Fri, Dec 18, 2009 at 6:59 AM, Bruce Nik brucev...@gmail.com wrote:
Hello Everyone,
I am making a simple index.php file which will allow a web user to enter
his $phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call
bridged. Following is the index.php and the contents of
extensions_custom.conf. When I submit the form nothing happens. I don't even
see Manager Connected msg. Your input will be much appreciated. I am
thinking I have some syntax problem.
/etc/asterisk/extensions_custom.conf
*[testphp]
exten = _X.,1,Answer()
exten = _X.,n,Dial(SIP/testTrunk/${EXTEN})
exten = _X.,n,Hangup()*
*
*
*
*
*
*
*
/var/www/html/clickncall/index.php
html
head
titleClicknCall/title
/head
bodybrbr
div align=center
?php
$sys_ip = 127.0.0.1;
$User_str = testphp;
$Secret_str = testphp;
if ($_POST['x']) {
$oSocket = fsockopen($sys_ip, 5038, $errnum, $errdesc) or die(Connection
to host failed);
fputs($oSocket, Action: login\r\n);
fputs($oSocket, Username: $User_str\r\n);
fputs($oSocket, Secret: $Secret_str\r\n\r\n);
fputs($oSocket, Events: off\r\n\r\n);
fputs($oSocket, Action: originate\r\n);
fputs($oSocket, Channel: SIP/testTrunk/$phoneNumb\r\n);
fputs($oSocket, Exten: $dialNumb\r\n);
fputs($oSocket, Context: testphp\r\n);
fputs($oSocket, Priority: 1\r\n\r\n);
fputs($oSocket, Timeout: 1\r\n);
fputs($oSocket, CallerId: $spoofNumb\r\n);
fputs($oSocket, Async: true\r\n);
fputs($oSocket, Action: Logoff\r\n\r\n);
fclose($oSocket);
print $_POST['x'];
} else {
print form method=\post\action=\$_SERVER[PHP_SELF]\;
print br;
print PHONE Number: input type=\text\name=\phoneNumb\;
print br;
print PARTY Number: input type=\text\name=\dialNumb\;
print br;
print FORGE Number: input type=\text\name=\spoofNumb\;
print br;
print input type=\Submit\ value=\ Dial
\;
print /form;
} ?
/div
/body
/html
*/etc/asterisk/manager_custom.conf*
[testphp]
secret = testphp
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
Thanks,
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[asterisk-users] Could Asterisk be crashing under high context switches?
Hello! I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks. We are an outgoing call center with 30 internal analog phones hooked up to 2 Rhino CB24 channel banks. The banks are connected to a Rhino R4T1 card in a Dell 2950 server with 8 gigs of RAM. The 2 other ports on the R4T1 go to our 2 PRIs. In this configuration, we have trouble maintaining stability. It may be fine for days, but soon the load slowly creeps up on the server from below 1 all the way up to 6 which is when no one can dial out and asterisk pretty much has to be killed to be stopped. We also have bandwidth.com set up as a SIP provider. If we use bandwidth.com, stability is greatly improved. I installed munin on the phone server yesterday and noticed something dramatic, I think! Asterisk became unstable 3 times yesterday. 2 of those times, the number of context switches went to almost 80k the first time, then over 70k the second. First question - is this abnormal for around 20 ongoing recorded calls? I did a little bit of searching and found this: http://wiki.sangoma.com/files/wanpipe-linux-asterisk-tutorials/How_to_Reduce_Asterisk_System_Loads.pdf It talks about zaptel/DAHDI chunk size and that directly affects system load. Second question - the document explains how to change the chunk size for Sangoma hardware. Is there a general way to do that for DAHDI? Thanks is advance! Jason Martin Metrix Matrix, Inc. 785 Elmgrove Rd, Bldg 1 Rochester, NY 14624 Office: 888-865-0065 x202 Mobile: 585-705-1400 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SAP-BCM Sip trunking
Hello, i have a problem with a Sip trunk to a SAP-BCM PBX. In and Outbound Calls works fine but when the SAP tries to transfer an inbound call to an outbound call there is no-way-audio. Two outbound calls could be transfered without any Problem. In the sip trace i see that the SAP BCM make something wrong with the Reinvite. (wrong SDP IP and Ports information, and only one reInvite package for the inbound call, nothing for the second) My Question is if there is someone who has a working trunk with a SAP BCM PBX and if they had the same problems. I´ve used Asterisk 1.2.35, 1.6.1.11 and also 1.6.2.rc8 but on every version it doenst work. Best regards Steve Smith ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Could Asterisk be crashing under high context switches?
On Fri, Dec 18, 2009 at 06:53, Jason Martin jmar...@metrixmatrix.comwrote: Hello! I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks. We are an outgoing call center with 30 internal analog phones hooked up to 2 Rhino CB24 channel banks. The banks are connected to a Rhino R4T1 card in a Dell 2950 server with 8 gigs of RAM. The 2 other ports on the R4T1 go to our 2 PRIs. In this configuration, we have trouble maintaining stability. It may be fine for days, but soon the load slowly creeps up on the server from below 1 all the way up to 6 which is when no one can dial out and asterisk pretty much has to be killed to be stopped. We also have bandwidth.com set up as a SIP provider. If we use bandwidth.com, stability is greatly improved. I installed munin on the phone server yesterday and noticed something dramatic, I think! Asterisk became unstable 3 times yesterday. 2 of those times, the number of context switches went to almost 80k the first time, then over 70k the second. First question - is this abnormal for around 20 ongoing recorded calls? I did a little bit of searching and found this: http://wiki.sangoma.com/files/wanpipe-linux-asterisk-tutorials/How_to_Reduce_Asterisk_System_Loads.pdf It talks about zaptel/DAHDI chunk size and that directly affects system load. Second question - the document explains how to change the chunk size for Sangoma hardware. Is there a general way to do that for DAHDI? Thanks is advance! Jason Martin Metrix Matrix, Inc. 785 Elmgrove Rd, Bldg 1 Rochester, NY 14624 Office: 888-865-0065 x202 Mobile: 585-705-1400 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Jason, Indeed what you are seeing is not typical. I don't have normal number available off-hand, but a system should have no problems whatsoever with 2 or 3 R4T1s. As you can expect, Rhino has thousands and thousands of customers running with no problem, which makes this instance the exception. All Rhino cards use the same amount of bus resources (time hold the PCI bus, data copied, etc) no matter if it's an R1T1 or R4T1, or how many active calls you have. There is no need to change the CHUNKSIZE as we have chosen the optimal solution (in our testing) for keeping system load down on hardware as minimal as a few hundred megahertz. That said, there's no way you can change the CHUNKSIZE on a Rhino card, it would require a completely different firmware. In my experience, I have seen issues similar to this arise from hard disk activity hogging the bus. Whether it's simultaneous recordings or perhaps a considerable amount of other reading/writing, what ends up happening is the CPU is switching between the Rhino card's interrupt and the IDE/SATA controller interrupt. When one of those interrupts becomes more frequent and holds the bus for too long, that takes time away from the R4T1 and data has to be discarded. We last saw these issues with nVidia hardware in the 2.6.9 kernels, but it's possible some derivative is affecting you. I would suggest investigating other factors that may be affecting system load when your call load increases. Context switches are simply a symptom and you still need to find the culprit. Regards, Bryce Chidester Rhino Equipment Corp. br...@rhinoequipment.com Tel: +1 (480) 621-4000, +1 (877) RHINO-T1 FAX: +1 (480) 961-1826 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wrapuptime?
As it is done today, the wrap-up time is not terribly useful in Asterisk, as it is fixed-length. If you need to implement it in a real-life scenario, it would be better to pause the agent when the call is through and have him unpause manually when he's done the wrap-up; this way you get a measurable metric as well. Just my two eurocents, l. 2009/12/18 Magnus Benngård magnu...@inputinterior.se Hi! Trying to understand how wrapuptime is working... I have written a small php script that let agents log in/out off a queue. That part is working as a clock but wrapuptime is not doing what I expect. Input Interiör - Queue Manager 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (5s holdtime, 94s talktime), W:0, C:8, A:1, SL:0.0% within 0s Members: SIP/0317998971 with penalty 2 (dynamic) (Not in use) has taken no calls yet SIP/0317998975 with penalty 2 (dynamic) (Not in use) has taken no calls yet SIP/0317998972 with penalty 1 (dynamic) (Not in use) has taken 8 calls (last was 2 secs ago) No Callers SIP/0317998972 did hang up 8 seconds ago, but if someone calls the queue at this moment, the call will start ringing on SIP/0317998972 again. I thought the wrapuptime should cause the call to go to one of the other agents. Did I miss anything in the configs or is it that we have different penalties or...? -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To Asterisk AMI Gurus - Tacking issue with originate
Thanks for the input.
Once the form is submitted the variables obtain a value which is the phone
number, dial number, and spoof number. I did the telnet and it works fine.
Now, I am really not a php coder and just stealing code from here and there
to make this working. Here is what I noticed that kind of works. If I do
this, I see an attempt to connect:
$oSocket = fsockopen($sys_ip, 5038, $errnum, $errdesc) or die(Connection to
host failed);
if ($_POST['x']) {
But I keep it like the normal way (connecting after form is submitted):
if ($_POST['x']) {
$oSocket = fsockopen($sys_ip, 5038, $errnum, $errdesc) or die(Connection to
host failed);
I am amazed that there is absolutely no proper documentation on how to
connect to Asterisk AMI with PHP. All tutuorial just mention: pass Action:
originate Channel: SIP/1234, blah blah blah and never give a simple example
of php.
Thanks a lot for your inputs guys.
On Fri, Dec 18, 2009 at 8:33 AM, Prince Singh pri...@drishti-soft.comwrote:
Obvious debugging steps at this point:-
1. Manually connecting to AMI and testing the commands
1. telnet 127.0.0.1 5038
2. Login command
3. Originate command
2. Use a dummy AMI listener instead of actual asterisk to see if the
events are going through
1. Stop asterisk (or change the manager port)
2. nc -l -k 127.0.0.1 5038
3. Confirm that 'nc' has taken the tcp port 5038
netstat -nlp | grep 5038
4. Test the php application to see if the login request can be seen
on the console from where 'nc' was started
3. Edit /etc/php.ini (or similar file) to enable logging and then check
the log file :)
On Fri, Dec 18, 2009 at 6:59 AM, Bruce Nik brucev...@gmail.com wrote:
Hello Everyone,
I am making a simple index.php file which will allow a web user to enter
his $phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call
bridged. Following is the index.php and the contents of
extensions_custom.conf. When I submit the form nothing happens. I don't even
see Manager Connected msg. Your input will be much appreciated. I am
thinking I have some syntax problem.
/etc/asterisk/extensions_custom.conf
*[testphp]
exten = _X.,1,Answer()
exten = _X.,n,Dial(SIP/testTrunk/${EXTEN})
exten = _X.,n,Hangup()*
*
*
*
*
*
*
*
/var/www/html/clickncall/index.php
html
head
titleClicknCall/title
/head
bodybrbr
div align=center
?php
$sys_ip = 127.0.0.1;
$User_str = testphp;
$Secret_str = testphp;
if ($_POST['x']) {
$oSocket = fsockopen($sys_ip, 5038, $errnum, $errdesc) or die(Connection
to host failed);
fputs($oSocket, Action: login\r\n);
fputs($oSocket, Username: $User_str\r\n);
fputs($oSocket, Secret: $Secret_str\r\n\r\n);
fputs($oSocket, Events: off\r\n\r\n);
fputs($oSocket, Action: originate\r\n);
fputs($oSocket, Channel: SIP/testTrunk/$phoneNumb\r\n);
fputs($oSocket, Exten: $dialNumb\r\n);
fputs($oSocket, Context: testphp\r\n);
fputs($oSocket, Priority: 1\r\n\r\n);
fputs($oSocket, Timeout: 1\r\n);
fputs($oSocket, CallerId: $spoofNumb\r\n);
fputs($oSocket, Async: true\r\n);
fputs($oSocket, Action: Logoff\r\n\r\n);
fclose($oSocket);
print $_POST['x'];
} else {
print form method=\post\action=\$_SERVER[PHP_SELF]\;
print br;
print PHONE Number: input type=\text\name=\phoneNumb\;
print br;
print PARTY Number: input type=\text\name=\dialNumb\;
print br;
print FORGE Number: input type=\text\name=\spoofNumb\;
print br;
print input type=\Submit\ value=\ Dial
\;
print /form;
} ?
/div
/body
/html
*/etc/asterisk/manager_custom.conf*
[testphp]
secret = testphp
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
Thanks,
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--
Regards,
Prince Singh
Drishti-Soft Solutions Pvt Ltd
62-A, First Floor,
Maruti Industrial Area,
Sector - 18, Gurgaon - 122016
Haryana, India.
P: 91 124 4771000
F: 91 124 4039120
W: http://www.drishti-soft.com
B: http://blog.drishti-soft.com
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Any views expressed in this message are those of the individual sender
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Re: [asterisk-users] Feature Request: GotoIfTimeWithOffset
On Friday 18 December 2009 04:11:36 Tzafrir Cohen wrote: On Thu, Dec 17, 2009 at 07:19:49PM -0600, Tilghman Lesher wrote: On Thursday 17 December 2009 17:23:22 Olivier wrote: When I was testing an IVR, I realized I miss a function I would call GotoIfTimeWithOffset. Today, this IVR is using function AEL GotoIfTime in several places. The problem is if it's 11pm at the moment I'm testing this IVR, I can't nicely test the 9am or 2pm branch. GotoIfTimeWithOffset would get 2 incoming arguments : - the first is a time range (just like GotoIfTime), - the second is a duration offset which you could delay or rewind time. After testing, you would just have to set this offset to 0, to get a production-ready dialplan, without changing a line. https://issues.asterisk.org/view.php?id=16464 I've made some important changes to your idea. The method for invoking this is setting a function to a particular date and time. The reason for this is to permit testing of a live system, without changing the actual dialplan. You could set up a test with a particular extension that sets this function before jumping to the incoming context, or you could use conditional evaluation, such as checking callerid for the administrator's cellphone number, in the incoming context to evaluate whether to set this function or not. Looks nice. Just one point: This is a global (or rather: channel-specific, but still common to all in thecontext of that channel) setting. Suppose I have ( and suppose Asterisk even accepts this syntax) : $(TESTTIME(...)} GotoIfTime(test1,...) GotoIfTime(test2,...) The syntax is actually: Set(TESTTIME()=2009-12-25 10:35:00 CST) Both test1 and test2 would be evaluated with the same offset as set in the TESTTIME() above. Is that what you want? I believe it is. Typically, people do multiple GotoIfTime's in a row, all predicated on several different conditions that they wish to evaluate. This is important, because a night message might say We will be open at 8:00, but on Christmas Day, they might wish the message to say instead We will not be open at all today. Order, then, is certainly important and worth testing. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HOW to record saynumber output
Hi all, the aims of this mail is to use saynumber fonctionality during Music On Hold while dialing. Music On Hold can only play a music file So Is there a way to pre-record the saynumber output and other .gsm file and then play the record file during Music On Hold ? all solutions are welcome regards Mickael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HOW to record saynumber output
If you have SOX, LAME and time, you can do about anything you want. The default moh files are wav, but a lot of folks use mp3 with the mpg123 player. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mickael ropars Sent: Friday, December 18, 2009 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] HOW to record saynumber output Hi all, the aims of this mail is to use saynumber fonctionality during Music On Hold while dialing. Music On Hold can only play a music file So Is there a way to pre-record the saynumber output and other .gsm file and then play the record file during Music On Hold ? all solutions are welcome regards Mickael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feature Request: GotoIfTimeWithOffset
On Fri, 18 Dec 2009, Tilghman Lesher wrote:
The syntax is actually:
Set(TESTTIME()=2009-12-25 10:35:00 CST)
1) Does this set the time to a fixed value or does it set the time at
the point of execution and then the clock increments from there?
2) Does this only affect gotoiftime() or does it affect every time value
associated with the executing channel? Would
exten = *,n,agi(foo,${EPOCH})
pass the number of seconds between 00:00:00 1970-01-01 UTC and
${TESTTIME}.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] To Asterisk AMI Gurus - Tacking issue with originate
El 18/12/09 11:31, Bruce Nik escribió: I am amazed that there is absolutely no proper documentation on how to connect to Asterisk AMI with PHP. All tutuorial just mention: pass Action: originate Channel: SIP/1234, blah blah blah and never give a simple example of php. http://phpagi.sourceforge.net/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HOW to record saynumber output
Hi Danny, I've already have a look to those tools, but I cannot see how I can record the saynumber output audio into a file Mickael 2009/12/18 Danny Nicholas da...@debsinc.com If you have SOX, LAME and time, you can do about anything you want. The default moh files are wav, but a lot of folks use mp3 with the mpg123 player. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *mickael ropars *Sent:* Friday, December 18, 2009 11:05 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] HOW to record saynumber output Hi all, the aims of this mail is to use saynumber fonctionality during Music On Hold while dialing. Music On Hold can only play a music file So Is there a way to pre-record the saynumber output and other .gsm file and then play the record file during Music On Hold ? all solutions are welcome regards Mickael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HOW to record saynumber output
Saynumber just does an EXECUTE BACKGROUND on the files in /var/lib/asterisk/sounds/digits. So to record a saynumber output of 23 to a moh file, you would do sox /var/lib/asterisk/sounds/digits/20.gsm /var/lib/asterisk/sounds/digits/3.gsm /var/lib/asterisk/moh/23.wav. If your moh processes randomly, the 23 would come up every x times. If you use classes to control moh, you can make it come up each time. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mickael ropars Sent: Friday, December 18, 2009 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HOW to record saynumber output Hi Danny, I've already have a look to those tools, but I cannot see how I can record the saynumber output audio into a file Mickael 2009/12/18 Danny Nicholas da...@debsinc.com If you have SOX, LAME and time, you can do about anything you want. The default moh files are wav, but a lot of folks use mp3 with the mpg123 player. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mickael ropars Sent: Friday, December 18, 2009 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] HOW to record saynumber output Hi all, the aims of this mail is to use saynumber fonctionality during Music On Hold while dialing. Music On Hold can only play a music file So Is there a way to pre-record the saynumber output and other .gsm file and then play the record file during Music On Hold ? all solutions are welcome regards Mickael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Waiting With Draytek ATA
Greetings all- I've got a rather odd situation and would like to know if anyone can shed some light on the issue. Some background- I've got an * system running 1.4.11 (yes I know it's older.. upgrades are planned at some point...). I also have a remote user with a cordless phone connected to a Draytek ATA device. When this user is on a call and receives another call via call waiting, they use the 'flash' button on their phone to switch to the other call. When this occurs, music on hold is started for the first call, and the second call is connected. However, at this point music on hold suddenly stops and audio from both calls can be heard together (and is rather garbled). Then, hitting flash again, call 2 is disconnected and call 1 is connected again. BUT, only one way audio(inbound to the user) is available on the first call now. I thought it could be a problem with MoH and ensured that was setup properly. Still the same problem. Then, I thought it could be a problem with the version of Asterisk I was running. As it turns out, a separate system running 1.2.13 works perfectly. So, at this point, I have to ask... are there any known issues like this that have been fixed in later versions than what I'm running? I know I'll probably receive a general blanket statement like upgrade to the latest but what I'm looking for is solid proof that an upgrade will fix it (something from the bug tracker maybe?). Or, maybe I'm going about this the wrong way and its something configured wrong elsewhere and * is not at fault? All thought and comments welcome. Thank you! Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feature Request: GotoIfTimeWithOffset
On Friday 18 December 2009 11:17:24 Steve Edwards wrote: On Fri, 18 Dec 2009, Tilghman Lesher wrote: The syntax is actually: Set(TESTTIME()=2009-12-25 10:35:00 CST) 1) Does this set the time to a fixed value or does it set the time at the point of execution and then the clock increments from there? It sets a fixed value. 2) Does this only affect gotoiftime() or does it affect every time value associated with the executing channel? Right now, it only affects GotoIfTime. Making it affect all places without making it global (which kind of defeats the purpose) might be rather difficult, as the channel reference is not handed down to every utility function involving time. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.0.20 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.0.20. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.0.20 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * clarify requirecalltoken option in iax.sample.conf (closes issue #16223), reported, patched by: bklang * Prevent double closing of FDs by EIVR (closes issue #16305), reported by: diLLec, patched, tested by: thedavidfactor * Fix multiple issues with musiconhold, which led to classes not getting destroyed properly. (closes issues #16279, #16207), reported by: parisioa, dcabot, patched by: tilghman, tested by: parisioa, tilghman * Send ack (response/message) after receiving manager action userevent (closes issue #16264), reported, patched by: dimas * Make manager response to Action: events finish with empty line (closes issue #16275), reported, patched by: vnovy This release also contains significant improvements to T.38 support. Anyone who has tried T.38 faxing in the past should try again as most problems should now be resolved. A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.20-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.20 Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.28 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.4.28. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.28 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Send ack (response/message) after receiving manager action userevent (closes issue #16264), reported, patched by: dimas * Do not modify the gain settings on data calls in chan_dahdi. (closes issue #15972), reported by: udosw, patched, tested by: alecdavis * fixes solaris segfault on dial with verbosity = 3 (closes issue #16193), reported by: asgaroth, patched by: snuffy, tested by: snuffy, asgaroth * fixes conditional jump or move depending on uninitialised STACK value (closes issue #16261), reported, patched by: edguy3 * Copy the peer CDR's userfield to the bridge CDR if it exists. (closes issue #14590), reported by: msetim, patched by Laureano, tested by: Laureano, mnicholson A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.4.28-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.28 Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.1.12 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.6.1.12. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.1.12 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix multiple issues with musiconhold, which led to classes not getting destroyed properly. (closes issues #16279, #16207), reported by: parisioa, dcabot, patched by: tilghman, tested by: parisioa, tilghman * Fix compatibility with valgrind 3.3 and older. (noticed in issue #16388), reported by: parisioa, patched by: atis, tested by: atis, parisioa * Prevent double closing of FDs by EIVR (closes issue #16305), reported by: diLLec, patched, tested by: thedavidfactor * Send ack (response/message) after receiving manager action userevent (closes issue #16264), reported, patched by: dimas * Make manager response to Action: events finish with empty line (closes issue #16275), reported, patched by: vnovy This release also contains significant improvements to T.38 support. Anyone who has tried T.38 faxing in the past should try again as most problems should now be resolved. A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.1.12-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.12 Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ringing for incoming call
Dear All, I am using Asterisk 1.4 on CentOS 5. I have an incoming DID provided by Vitelity. When the number is called it goes to my Asterisk box. The protocol is SIP. This all works just fine if I answer the call and begin a playback. I want to let the number ring for a few seconds before it is answered, and would like the caller to hear it ringing. I have tried: ... exten = s,n,Answer exten = s,n,Playtones(ring) exten = s,n,Wait(10) exten = s,n,StopPlaytones() exten = s,n,BackGround(sound file) ... also ... exten = s,n,Answer exten = s,n,Ringing() exten = s,n,Wait(10) exten = s,n,BackGround(sound file) ... I have also tried moving the Answer app to right before the BackGround app. In all cases when I call the number I never hear it ringing. After the 10 second delay, the BackGround app does run. Connecting to the CLI does not give me any useful information - for example the Ringing app is shown to run, but the caller does not hear it. Any suggestions? Many thanks! -- Bob Smither smit...@c-c-i.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6.2.0 Now Available!
The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and Asterisk-Addons 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.0 is the first feature release since Asterisk 1.6.1.0, which was released April 27, 2009. Many new features have been included in this release. For a complete list of changes, please see the CHANGES file. For those upgrading from a previous release, please see UPGRADE.txt It should be explicitly stated that Asterisk 1.6.2.0 is a major upgrade over any previous release, and special care should be taken when upgrading existing systems. Please see the UPGRADE.txt file for more information, available at: http://svn.asterisk.org/svn/asterisk/tags/1.6.2.0/UPGRADE.txt A detailed overview to the new features available in Asterisk 1.6.2.0 are forthcoming within the next few days. Please watch http://blogs.asterisk.org for further information! Below is a summary of several new features available in this release: * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with support for LibOpenR2. http://www.libopenr2.org/ * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this option is enabled, Asterisk will watch for a CNG tone in the incoming audio for a received call. If it is detected, the channel will jump to the 'fax' extension in the dialplan. * A new application, Originate, has been introduced, that allows asynchronous call origination from the dialplan. * Added ConfBridge dialplan application which does conference bridges without DAHDI. For information on its use, please see the output of core show application ConfBridge from the CLI. * extensions.conf now allows you to use keyword same to define an extension without actually specifying an extension. It uses exactly the same pattern as previously used on the last exten line. For example: exten = 123,1,NoOp(something) same = n,SomethingElse() * Asterisk now provides the ability to define custom CLI aliases. For example, if you would like to define short form aliases for frequently used commands, such as sh ch for core show channels, that is now possible. See the cli_aliases.conf configuration file for more information. * Asterisk now has support for subscribing to the state of remote voice mailboxes via SIP. * Asterisk now includes expanded HD codec support. G.722.1 and G.722.1C (Siren7/Siren14) passthrough, recording, and playback is now supported. Transcoding will be made available via add-on modules soon for this version of Asterisk. This is just a subset of the changes available in this release. Please see the CHANGES file for additional information, available at: http://svn.asterisk.org/svn/asterisk/tags/1.6.2.0/CHANGES A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.2.0-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.0 For more information about Asterisk-Addons 1.6.2.0, please see the ChangeLog available at: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-addons-1.6.2.0 Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.12 Now Available
Is the new Fax For Asterisk being released in conjunction with this release? Thanks, --Warren Selby On Dec 18, 2009, at 4:59 PM, Asterisk Development Team asteriskt...@digium.com wrote: The Asterisk Development Team has announced the release of Asterisk 1.6.1.12. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.1.12 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix multiple issues with musiconhold, which led to classes not getting destroyed properly. (closes issues #16279, #16207), reported by: parisioa, dcabot, patched by: tilghman, tested by: parisioa, tilghman * Fix compatibility with valgrind 3.3 and older. (noticed in issue #16388), reported by: parisioa, patched by: atis, tested by: atis, parisioa * Prevent double closing of FDs by EIVR (closes issue #16305), reported by: diLLec, patched, tested by: thedavidfactor * Send ack (response/message) after receiving manager action userevent (closes issue #16264), reported, patched by: dimas * Make manager response to Action: events finish with empty line (closes issue #16275), reported, patched by: vnovy This release also contains significant improvements to T.38 support. Anyone who has tried T.38 faxing in the past should try again as most problems should now be resolved. A summary of changes in this release can be found in the release summary: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.1.12-summary.txt For a full list of changes in this releases, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ ChangeLog-1.6.1.12 Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.12 Now Available
Warren Selby wrote: Is the new Fax For Asterisk being released in conjunction with this release? If it's not already available, then it will be available very early next week. Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.12 Now Available
How does Fax for Asterisk work? On Fri, Dec 18, 2009 at 7:51 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: Warren Selby wrote: Is the new Fax For Asterisk being released in conjunction with this release? If it's not already available, then it will be available very early next week. Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.12 Now Available
http://www.digium.com/en/products/software/faxforasterisk.php Thanks, --Warren Selby On Dec 18, 2009, at 7:11 PM, Thomas Perron thomas.per...@gmail.com wrote: How does Fax for Asterisk work? On Fri, Dec 18, 2009 at 7:51 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: Warren Selby wrote: Is the new Fax For Asterisk being released in conjunction with this release? If it's not already available, then it will be available very early next week. Leif Madsen. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing for incoming call
Try putting the wait before the Answer. ... exten = s,n,Wait(10) exten = s,n,Answer ... On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither smit...@c-c-i.com wrote: Dear All, I am using Asterisk 1.4 on CentOS 5. I have an incoming DID provided by Vitelity. When the number is called it goes to my Asterisk box. The protocol is SIP. This all works just fine if I answer the call and begin a playback. I want to let the number ring for a few seconds before it is answered, and would like the caller to hear it ringing. I have tried: ... exten = s,n,Answer exten = s,n,Playtones(ring) exten = s,n,Wait(10) exten = s,n,StopPlaytones() exten = s,n,BackGround(sound file) ... also ... exten = s,n,Answer exten = s,n,Ringing() exten = s,n,Wait(10) exten = s,n,BackGround(sound file) ... I have also tried moving the Answer app to right before the BackGround app. In all cases when I call the number I never hear it ringing. After the 10 second delay, the BackGround app does run. Connecting to the CLI does not give me any useful information - for example the Ringing app is shown to run, but the caller does not hear it. Any suggestions? Many thanks! -- Bob Smither smit...@c-c-i.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To Asterisk AMI Gurus - Tacking issue with originate
Thanks for the input. I have explored phpagi from sourceforge but they don't have any documentation or it's poor. I am just wondering if any can contribute their working php file with me that does Originate action. Thanks On Fri, Dec 18, 2009 at 12:24 PM, Alex Villacís Lasso a_villa...@palosanto.com wrote: El 18/12/09 11:31, Bruce Nik escribió: I am amazed that there is absolutely no proper documentation on how to connect to Asterisk AMI with PHP. All tutuorial just mention: pass Action: originate Channel: SIP/1234, blah blah blah and never give a simple example of php. http://phpagi.sourceforge.net/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.0 Now Available!
At 03:14 PM 12/18/2009, you wrote: The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and Asterisk-Addons 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ And any plan for Skype for Asterisk? Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing for incoming call
On Fri, 2009-12-18 at 19:58 -0600, Steve Johnson wrote: Try putting the wait before the Answer. ... exten = s,n,Wait(10) exten = s,n,Answer ... Thanks Steve. I tried that: On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither smit...@c-c-i.com wrote: Dear All, snip ... exten = s,n,Answer exten = s,n,Ringing() exten = s,n,Wait(10) exten = s,n,BackGround(sound file) ... I have also tried moving the Answer app to right before the BackGround app. snip i.e., after the Wait, but still no joy. Anything else I need to look at? Thanks, -- Bob Smither smit...@c-c-i.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.0 Now Available!
On Fri, 2009-12-18 at 19:43 -0800, Ira wrote: At 03:14 PM 12/18/2009, you wrote: The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and Asterisk-Addons 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ And any plan for Skype for Asterisk? Ira http://www.digium.com/en/products/software/skypeforasterisk.php (but it is not free) -- Bob Smither smit...@c-c-i.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.0 Now Available!
At 08:31 PM 12/18/2009, you wrote: On Fri, 2009-12-18 at 19:43 -0800, Ira wrote: At 03:14 PM 12/18/2009, you wrote: The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and Asterisk-Addons 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ And any plan for Skype for Asterisk? Ira http://www.digium.com/en/products/software/skypeforasterisk.php (but it is not free) I know it's not free, I want it for 1.6.2 which I've been running for quite a while and it's not there for download yet. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing for incoming call
If you try just this, what does the caller hear? It should be ringing for the first 20 sec, and then maybe the congestion tone afterwards. exten = s,1,Wait(20) exten = s,n,Hangup You shouldn't need/use the Ringing() command at all, as the initial ring before your system answers would be generated by the provider. If wait ... answer doesn't work for you, you'll have to provide more output from the CLI and tell us more about your configuration. On Fri, Dec 18, 2009 at 10:29 PM, Bob Smither smit...@c-c-i.com wrote: On Fri, 2009-12-18 at 19:58 -0600, Steve Johnson wrote: Try putting the wait before the Answer. ... exten = s,n,Wait(10) exten = s,n,Answer ... Thanks Steve. I tried that: On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither smit...@c-c-i.com wrote: Dear All, snip ... exten = s,n,Answer exten = s,n,Ringing() exten = s,n,Wait(10) exten = s,n,BackGround(sound file) ... I have also tried moving the Answer app to right before the BackGround app. snip i.e., after the Wait, but still no joy. Anything else I need to look at? Thanks, -- Bob Smither smit...@c-c-i.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.0 Now Available!
Ira schrieb: At 08:31 PM 12/18/2009, you wrote: On Fri, 2009-12-18 at 19:43 -0800, Ira wrote: At 03:14 PM 12/18/2009, you wrote: The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and Asterisk-Addons 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ And any plan for Skype for Asterisk? Ira http://www.digium.com/en/products/software/skypeforasterisk.php (but it is not free) I know it's not free, I want it for 1.6.2 which I've been running for quite a while and it's not there for download yet. I've been running SFA on 1.6.2.x-rcX for quite some time with no stability issues at all. Did you give it a try? Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
