Re: [Asterisk-Users] ACD Queues Agent Status
What I had to do to make this work was use global variables.
First I declared:
QUEUEAGENTSUPPORT=0
Then in the agent logon I used (for logoff use - 1)
exten = 81,2,SetGlobalVar(QUEUEAGENTSUPPORT=$[${QUEUEAGENTSUPPORT} + 1])
so ${QUEUEAGENTSUPPORT} will contain the number of agents logged into that
queue.
Now you can GotoIf(${QUEUEAGENTSUPPORT} 0..
Regards,
Jim
On Tue, 11 Jan 2005, Ronald Hartmann wrote:
Good Day List,
I am finalizing my research on the ACD Ques and have (what I
hope to be) one last hurdle.
Is there any feasible way to determine if a que has agents
currently available to take a call.
I have looked at the Show Queues, show queue quename, show
agents and they all give me pieces,
However, for example lets say that Agent-1, and Agent-2 belong
to Group1.
When I perform show queue myqueue it only shows me that
Agent/@1 is logged in... it does not tell me
Which agents are in that group, nor does it tell me if they are
accepting calls.
Further, is there a way to determine if an agent is in the
WrapUptime Mode?
Any url links would be helpful.
Ultimately what I am looking to do is as follows.
In my dialplan, I want to be able to check for agent
availability, and then play a your being placed in que,
If there are no agents immediately available to take a call.
Then pass the call to the Queue() application.
However, if there is an agent immediately available to take a
call I will not play a file.
Example:
[que-test]
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = s,5, IF AGENTS ARE AVAILBLE TO TAKE CALL
GOTO6 ELSE GOTO5
exten = s,5,Background(your call is being placed in
the Support Que please hold for an agent)
exten = s,6,Queue(techsupport)
exten = s,7,Playback(All Agents are busy, please leave
your name and callback number)
exten = s,8,VoiceMail(SUPPORT_MAILBOX) ;
Final Thought,
Is it possible to Set the Que up such that it plays a special
recording every X interval.
Currently it will play musiconhold,
Then at specific intervals it can tell the caller where they are
in the que and hold times.
I would like to be able to Also at certain intervals play a
recording stating.
If you would like to leave a call back number please press X
now.
Etc.
Thanks for your time in reading this note. I am sure that as soon as
I send this it will turn into a RTFM dummy
But I have googled and wikied my fingers to the bone and now require
some outside direction.
Ron.
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==
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==
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Re: [Asterisk-Users] fax e-mail spandsp
Basically the changes in the apps/Makefile have progressed while the patch
makefile have not. Here is a current patch that works as of
CVS-HEAD-01/06/05-14:47:06
Regards,
Jim
On Fri, 7 Jan 2005, Altus Snyman wrote:
I'm trying to install spandsp
But when I try to patch the Makefile it gives this error
[EMAIL PROTECTED] apps]# patch apps_makefile.patch
patching file Makefile
Reversed (or previously applied) patch detected! Assume -R? [n] y
Hunk #1 succeeded at 41 (offset -6 lines).
Hunk #2 FAILED at 67.
is it ok to go on
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==
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http://www.jimradford.com/
==--- Makefile.orig 2005-01-05 17:00:52.0 -0800
+++ Makefile2005-01-05 17:02:09.0 -0800
@@ -49,6 +49,8 @@
APPS+=$(shell if [ -f /usr/include/linux/zaptel.h ]; then echo app_zapras.so
app_meetme.so app_flash.so app_zapbarge.so app_zapscan.so ; fi)
APPS+=$(shell if [ -f /usr/local/include/zaptel.h ]; then echo app_zapras.so
app_meetme.so app_flash.so app_zapbarge.so app_zapscan.so ; fi)
APPS+=$(shell if [ -f /usr/include/osp/osp.h ]; then echo app_osplookup.so ;
fi)
+APPS+=$(shell if [ -f /usr/include/spandsp.h ]; then echo app_rxfax.so
app_txfax.so ; fi)
+APPS+=$(shell if [ -f /usr/local/include/spandsp.h ]; then echo app_rxfax.so
app_txfax.so ; fi)
CURLLIBS=$(shell curl-config --libs)
ifneq (${CURLLIBS},)
@@ -78,6 +80,12 @@
rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so
rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so
+app_rxfax.so : app_rxfax.o
+ $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
+
+app_txfax.so : app_txfax.o
+ $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
+
app_curl.so: app_curl.o
$(CC) $(SOLINK) -o $@ $ $(CURLLIBS)
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[Asterisk-Users] lockup problem with inbound iax calls
I've ran into a strange problem Using current HEAD version (CVS-HEAD-12/22/04-20:45:05), Whenever anyone calls in to my FWD number it locks the machine up good. If I go back to the stable version (CVS-v1-0-12/23/04-07:05:33) It works fine. I've recompiled the current HEAD version (make clean), even re-downloaded a fresh copy from CVS with the same results. When I compile and install the STABLE version it seems to work fine. What sort of additional information can I provide for debugging? Regards, Jim -- == Jim Radford [EMAIL PROTECTED] http://www.jimradford.com/ == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Goto and exten = syntax
GoTo(6275,1) or (6275|1) are the same, and that would be goto(extension,priority) in the same context as you've said. Regards, Jim On Thu, 23 Dec 2004, Dorn Hetzel wrote: I understand some of the basic Goto() forms, such as Goto(context,extension,priority) and Goto(extension,priority) [within context I presume]. Can someone Explain Goto(6275|1) as found in the sample extensions.conf? Is this the same as Goto(6275,1) just with a different delimiting character? In a use like: exten = s,1,Wait,2 exten = s,n(dial),Dial(SIP/sip1,20,tr) two questions: does the n mean priority value of the previous line + 1 ? does the (dial) mean create a variable? named dial which can be referenced as priority in a Goto to get to this line? I looked at the Wiki but couldn't determine these points for sure... Regards, -Dorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- == Jim Radford [EMAIL PROTECTED] http://www.jimradford.com/ == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Status of Queue?
Is there a way of getting the status of a Queue before sending a caller into it? -- == Jim Radford [EMAIL PROTECTED] http://www.jimradford.com/ == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJPhone SIP Tab
On Sat, 18 Dec 2004, Norman Zhang wrote: Thats someone elses instructions. I did add a link under the Please Note section however. Regards, Jim Jim Radford wrote: I've just created a SJPhone page with screen shots for those new to asterisk or anyone trying to get the more current version of SJPhone working. Feel free to send me any feedback directly. http://www.jimradford.com/asterisk/sjphone/ Thanks for the excellent reference Jim. Could you update your page at http://www.voip-info.org/wiki-Asterisk+phone+sjphone too? Regards, Norman Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- == Jim Radford [EMAIL PROTECTED] http://www.jimradford.com/ == ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJPhone - Proxy Authentication Required
Grady: This might help: http://www.jimradford.com/asterisk/sjphone/ Regards, Jim On Sat, 18 Dec 2004, Grady Trew, Jr. wrote: Greetings... I'm really having a problem getting a SIP client setup on my end. I keep getting the Proxy Authentication Required popup dialog from SJPhone when the number finishes dialing out. It really doesn't matter what the number is or if it goes out IAX, PSTN, Local Extension, etc. The CLI debug info below is of a 800 call routed via iaxtel, but is the typical thing I see. I also included my sip.conf too. Any suggestions would be appreciated. I suspect it's something glaring me right in the face. Thanks for your help in advance... and happy holidays! Grady SJphone - Error Popup Dialog -- Number not available Call rejected: 407 Proxy Authentication Required From Asterisk CLI (asterisk -rdc) - -- Registered SIP 'grady' at 67.67.228.3 port 58877 expires 120 -- Executing Dial(SIP/grady-5f12, IAX2/[EMAIL PROTECTED]/18005551212) in new stack -- Called [EMAIL PROTECTED]/18005551212 -- Call accepted by 69.73.19.178 (format gsm) -- Format for call is gsm Dec 18 14:41:11 WARNING[5660]: chan_iax2.c:5967 socket_read: Received mini frame before first full voice frame Dec 18 14:41:11 WARNING[5660]: chan_iax2.c:5967 socket_read: Received mini frame before first full voice frame Dec 18 14:41:15 WARNING[5658]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Non-critical Response) -- IAX2/iaxtel/1 answered SIP/grady-5f12 Dec 18 14:41:21 WARNING[5658]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 2 (Non-critical Response) Dec 18 14:41:22 WARNING[5658]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 2 (Non-critical Response) Dec 18 14:41:22 WARNING[5658]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 2 (Non-critical Response) Dec 18 14:41:30 WARNING[5658]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 2 (Non-critical Response) Dec 18 14:41:30 WARNING[5658]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 2 (Non-critical Response) Dec 18 14:41:46 WARNING[5658]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 2 (Non-critical Response) Dec 18 14:41:46 WARNING[5658]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 2 (Non-critical Response) -- Hungup 'IAX2/iaxtel/1' == Spawn extension (office, 18005551212, 1) exited non-zero on 'SIP/grady-5f12' sip.conf - [general] context=default port=5060 bindaddr=0.0.0.0 allow=all nat=no localnet=192.168.0.0/255.255.0.0 localnet=10.0.0.0/255.0.0.0 localnet=172.16.0.0/12 localnet=169.254.0.0/255.255.0.0 [grady] type=friend context=office host=dynamic dtmfmode=rfc2833 username=grady secret=gt1964 canreinvite=no reinvite=no callerid=Grady Trew 302 allow=all ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- == Jim Radford [EMAIL PROTECTED] http://www.jimradford.com/ == ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJPhone SIP Tab
I've just created a SJPhone page with screen shots for those new to asterisk or anyone trying to get the more current version of SJPhone working. Feel free to send me any feedback directly. http://www.jimradford.com/asterisk/sjphone/ Regards, Jim On Fri, 17 Dec 2004, [EMAIL PROTECTED] wrote: So for newby users of SJPhone... can you tell us exactly what goes in what box to connect to a standard AsteriskPBX using the latest interface. I've had no luck so far. thanks... On Wed, 2004-12-08 at 09:40, Girish Gopinath wrote: Hi, --- Norman Zhang [EMAIL PROTECTED] wrote: I'm following, http://www.voip-info.org/wiki-Asterisk+phone+sjphone. However, I cannot find the SIP tab. Would someone please give me a few pointers? The screen capture can be seen at URL below The wiki talks about an older version of SJPhone. The one you are using is the latest version (with stun support). Click on the profiles tab and create a new profile. You'll get an options dialog in which you can set the values. Regards, Girish __ Do you Yahoo!? Yahoo! Mail - Easier than ever with enhanced search. Learn more. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users P ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- == Jim Radford [EMAIL PROTECTED] http://www.jimradford.com/ == ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI Helpdesk/Trouble Ticketing application
Greetings All, I've created a sample trouble ticketing management script in perl that allows the management of trouble tickets, call routing, etc that interfaces with the perldesk helpdesk application. It is mainly for an example of what integration between asterisk and real world apps is possible. Basic instructions are included in the script itself including instructions for accessing a live demo, feel free to beat the hell out of it. Download From: http://www.jimradford.com/asterisk/perldesk.agi Enjoy! Regards, Jim -- == Jim Radford [EMAIL PROTECTED] http://www.jimradford.com/ == ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk's Empty Folder
You need to do a: make install and then make samples to install sample conf files. Jim On Wed, 8 Dec 2004, Adnan Ahmed wrote: Hello *'s, I have recently installed CentOS v3.3 and i have latest stable Asterisk's source code ,i compiles it shows no error but when i am looking for sip.conf,zapata.conf ,i am amazed the /etc/asterisk folder was empty i am compling several times but no luck what's the problem i compiled in the order of zaptel,libpri,asterisk. I send some traces of my asterisk's compilation kindly pointout my mistakes. o synths.o synths.c synths.c:172: warning: no previous prototype for `synths_' synths.c: In function `synths_': synths.c:401: warning: implicit declaration of function `irc2pc_' synths.c:402: warning: implicit declaration of function `bsynz_' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I ../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-v1-0-12/08/04-14:03:10\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\ when i compile asterisk these errors are aoming several times in several files and at the end + Asterisk Installation Complete ---+ +YOU MUST READ THE SECURITY DOCUMENT+ + Asterisk has successfully been installed. + + If you would like to install the sample + + configuration files (overwriting any + + existing config files), run: + + make samples+ kindly pointout what's wrong i am doing bocz i spend almost a day or above but all in vein. Thanks in Advance Adnan Ahmed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- == Jim Radford [EMAIL PROTECTED] http://www.jimradford.com/ == ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
