[Asterisk-Users] Queue Optional URL Problem
Dear All, Please help, I've wondering if there's any softphone run on Linux base will accept Queue's optional URL? I've been tried using the Zulys's LIP4Z and kPhone, after incoming calls accpted, it's nothing happen. Although I've checked with the Queue Log, the URL parameter is also sent.. Thanks! Regards, R Wong___ 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] Analog to Digital
Dear All, Current my client having total 32 POTS line and they want to change to an SoftPBX system. Is there any convertor can convert Analog-to-Digital? Since they don't even want to change to E1 digital line! Thanks for your advise first! Regards, R Wong ___ 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] BillSec and CLID in CDR Problem
Dear All, I am not familer to the C Language and I am not sure what's happening... After upgraded the Asterisk to CVS with header "CVS-HEAD-10/25/04-02:20:29", the CDR record of BillSec and CLID seems to be incorrect. Before upgrade, I can using the SetCallerID to make the CLI to the number that the system auto dialed out. But now it didn't change. For the billsec, before upgrade if caller hangup, the billsec is also update but now it only remains "0" unless it goes into the Queue. Is there any modification I can make to the configuration or Makefile to make it working again? Thanks for your help!!! Regards, R Wong ___ 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] Reload cause Sound Volumn becomes very loud
Hi all, I am running the Asterisk with CVS-HEAD-10/25/04. When I type reload in console, whatever the incoming/outgoing sound volumn becomes very loud until I stop the asterisk and restart it. It's running no problem before I've upgrade the asterisk. Is there any configuration I need to modify? Since I'm not able to find any information in this list nor voip-info.org (maybe I've overlook, if so please point me to the correct URL..) Thanks! Regards, R Wong ___ 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] No sound from IVR scripts, yet calls placed without any problem.
not sure if this can help since I'm also a beginner to asterisk... how about the sound file location? i've got an experience IF you are not using the english, you need to let asterisk know the file location. for example, you've got a file named "testing.wav(gsm)" and it located in /var/lib/asterisk/sounds/xx1 in the extensions.conf, you need to place something like "xx1/testing.wav(gsm)". Regards, R Wong - Original Message - From: "Nick Barnes" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]> Sent: Saturday, September 18, 2004 1:11 AM Subject: [Asterisk-Users] No sound from IVR scripts,yet calls placed without any problem. > > Hi, > > Bit of a puzzle this one - let's see if anybody else can shed some light... > > I Ghosted my Asterisk box to build one for a colleague. > I added one HFC card to make the total two and amended zaptel.conf and > zapata.conf accordingly. > I tested it with my handsets on my ISDN lines and it worked AOK. > I unplugged the box and took it to his house and plugged it in. > He doesn't have any ISDN lines installed as yet (they're coming next week). > He can make calls between extensions with no problem at all. > If he dials a number which extensions.conf passes to an IVR script, the > console shows what's going on (i.e. playing this, playing that, playing the > other, waiting for digit, got digit 7, playing somethingelse...), but > nothing is heard at the handset. > I've disabled zap stuff, removed the modules and it still does the same > thing. > I've been through every file and every configuration option and I can't work > out why it did work and now it doesn't. My first thought was some sort of > firewall was blocking things, but it's all on one LAN with no intelligent > switches or anything else - it should just work. > > Does anybody have any ideas? > > Asterisk CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a > > Nick Barnes > > > > > ___ > 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 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] Call Transfer
Dear All, How can I make a call transfer and line release after connected? I've found the Transfer(Zap/..) is not working as expect Thanks for your help! regards, R Wong ___ 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 didn't get var from Asterisk?
Dear All,
Just hope your guys out there
can help me through..since I've been playing for serval hoursand still not
able to resolve it...
The workflow as I've created an
.call file for Asterisk to call out and it's working fine with outdial, passing
variable to asterisk..But the problem is when the calls reached Context and
execute AGI script, the script didn't get any variable from the
agi...
The .call as the following
(I'm using the PHP to generate the call file) :
$channel="Channel:
Zap/g2/\r\n";
$channel.="Callerid: \r\n";
$channel.="Context:
autodial-agent\r\n";
$channel.="Extension: s\r\n";
$channel.="Priority: 1\r\n";
$channel.="SetVar:
AMSISDN=xxx\r\n";
$channel.="SetVar:
ADESKNUMBER=\r\n";
$channel.="SetVar:
ASTAFFNO=\r\n";
$channel.="SetVar: ADBID=xxx\r\n";
$channel.="MaxRetries: 1\r\n";
$channel.="RetryTime: 90\r\n";
$channel.="WaitTime: 60\r\n";
The AGI Script:
#!/usr/bin/php -q
require ("phpagi.php"); $agi=new
AGI();
$dbid=$agi->get_var("ADBID");
$tt=$agi->set_var("Adbid1", $dbid); echo "VERBOSE DID:
$dbid"; ?>
Console:
-- Launched AGI Script
UpdateAutoDial.php UpdateAutoDial.php|45:
'agi_request' => 'UpdateAutoDial.php'
UpdateAutoDial.php|45: 'agi_channel' => 'Zap/1-1'
UpdateAutoDial.php|45: 'agi_language' => 'en'
UpdateAutoDial.php|45: 'agi_type' => 'Zap'
UpdateAutoDial.php|45: 'agi_uniqueid' =>
'1095233534.12' UpdateAutoDial.php|45:
'agi_callerid' => '' UpdateAutoDial.php|45:
'agi_dnid' => '' UpdateAutoDial.php|45:
'agi_rdnis' => 'unknown' UpdateAutoDial.php|45:
'agi_context' => 'autodial-agent'
UpdateAutoDial.php|45: 'agi_extension' => 's'
UpdateAutoDial.php|45: 'agi_priority' => '3'
UpdateAutoDial.php|45: 'agi_enhanced' => '0.0'
UpdateAutoDial.php|45: 'agi_accountcode' => ''
UpdateAutoDial.php|45: ';dig' => '/usr/bin/dig'
UpdateAutoDial.php|45: 'debug' => 'true'
UpdateAutoDial.php|45: >> GET VARIABLE ADBID
UpdateAutoDial.php|45: >> SET VARIABLE Adbid1
UpdateAutoDial.php|45: DID: -- AGI
Script UpdateAutoDial.php completed, returning 0
extension:
[autodial-agent]
exten =>
s,1,SetVar(agi_did=${ADBID}) exten =>
s,2,SetVar(DMSISDN=${AMSISDN}) exten =>
s,3,AGI(UpdateAutoDial.php|${ADBID}) exten =>
s,4,NoOp(DBID1: ${ADBID}) exten =>
s,5,Hangup
Thanks in advances for your
help!
Regards,
R Wong
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[Asterisk-Users] Call Queues, CallerID, SIP and AutoDial
Hello, Current moment, I've successfully put the incoming calls into Queues and dial to an idle agents. When the agents answer the calls, the agents can hear the pre-recorded message to incidate what's the service that the call is calling. But there one problem that I'm not able to make it having the Caller ID display on the X-Lite. Even I try to make a call direct transfer from Asterisk to SIP X-Lite, it's not displaying the CallerID too! Is there's any method I can show the CallerID on the X-Lite? By the way, I also want to know how can I make the Auto Dial depends on current idle agents? I know that when I put the .call files into /var/spool/asterisk/outgoing, the Asterisk will make outbound call regardness how many files there...any suggestion? Thanks in advance! Regards, R Wong ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error Compiling MySQL Friends
Hi, try issue: export lang=C before make. Regards, R. Wong - Original Message - From: "imail" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, August 26, 2004 11:13 PM Subject: Re: [Asterisk-Users] Error Compiling MySQL Friends > Thanks Flynn but still no go. > I looked at line 141 and it seems to be fine. > > elifeq ($(USE_SIP_MYSQL_FRIENDS),1) > > I also tried removing the comma, and putting in a tab but I get the same > error. > I havent made any changes to the file, it was downloaded automatically via > cvs. > Any other thoughts? > > - Original Message - > From: "el Flynn" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[EMAIL PROTECTED]> > Sent: Thursday, August 26, 2004 1:45 AM > Subject: Re: [Asterisk-Users] Error Compiling MySQL Friends > > > > imail wrote: > > > All, > > > I edited the Makefile under asterisk/channels and set: > > > > > > USE_MYSQL_FRIENDS=1 > > > USE_SIP_MYSQL_FRIENDS=1 > > > > > > When I do a make clean ; make install > > > I get the following > > > > > > for x in res channels pbx apps codecs formats agi cdr astman stdtime; do > > > make -C $x clean || exit 1 ; done > > > make[1]: Entering directory `/usr/src/asterisk/res' > > > rm -f *.so *.o .depend > > > make[1]: Leaving directory `/usr/src/asterisk/res' > > > make[1]: Entering directory `/usr/src/asterisk/channels' > > > Makefile:141: *** missing separator. Stop. > > > make[1]: Leaving directory `/usr/src/asterisk/channels' > > > make: *** [clean] Error 1 > > > > Could it be that your problem is coming from the error: > > > >Makefile:141: *** missing separator. Stop. > > > > From the rest of the output it doesn't seem to imply there's something > > missing where MySQL is concerned. Googling on "makefile" and "missing > > separator" gave me this link that may be of help: > > > > http://www.cygwin.com/ml/cygwin/2003-07/msg00341.html > > > > Although I may be way off, you could try that out first and see if it > > doesn't solve the problem. > > > > Cheers, > > Flynn > > > > ___ > > 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 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 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] Out Dial Problem
e we're zombie or need a soft hangup: c0=SIP/2000-e12c, c1=Zap/17- 1, flags: No,No,No,Yes Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:2679 ast_channel_bridge: Bridge stops bridging channels SIP/2000-e12c and Zap/17-1 Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:733 ast_hangup: Hanging up channel 'Zap/17-1' Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:1902 zt_hangup: zt_hangup (Zap/17-1) Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2417 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/17-1 Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:1930 zt_hangup: Hangup: channel: 17 index = 0, normal = 38, callwait = -1, thirdcall = -1 Urgent handler Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2066 zt_hangup: Not yet hungup... Calling hangup once with icause, and clearing call Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2329 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/17-1 Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:1151 update_conf: Updated conferencing on 17, with 0 conference users Aug 26 15:54:17 DEBUG[-1260983376]: chan_zap.c:2411 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/17-1 Urgent handler -- Hungup 'Zap/17-1' Urgent handler Aug 26 15:54:17 DEBUG[-1260983376]: app_dial.c:974 dial_exec: Exiting with DIALSTATUS=ANSWER. Aug 26 15:54:17 DEBUG[-1260983376]: pbx.c:1827 ast_pbx_run: Spawn extension (from-sip,008522112,3) exitednon-zero on 'SIP/2000-e12c' Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:733 ast_hangup: Hanging up channel 'SIP/2000-e12c' Aug 26 15:54:17 DEBUG[-1260983376]: chan_sip.c:1717 sip_hangup: sip_hangup (SIP/2000-e12c) Aug 26 15:54:17 DEBUG[-1260983376]: chan_sip.c:1732 sip_hangup: update_user_counter(2000) - decrement inUse counter Aug 26 15:54:17 DEBUG[-125376]: chan_sip.c:817 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found And I'm not sure what's happening which the call actually didn't dial out..I hope someone out there can help me in... Thanks! R. Wong The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. ___ 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
