[asterisk-users] Festival Not Working
Hi, I have been fighting with this for days now. I can't seem to get festival to work with asterisk. I have followed the wiki on setting up asterisk to work with festival but no worky. I can run l the festival client and that works just fine but when I try to do it from asterisk I get the same problmes that are on the wiki. Here is how my setup is. Running Debian 3.1 asterisk 1.2.9 festival 1.95 I am using the default /etc/asterisk/festival.conf I have added in the specified changes into festival.scm Here is what shows in my festival-server.log file server Fri Aug 4 11:20:13 2006 : Festival server started on port 1314 client(1) Fri Aug 4 11:20:17 2006 : accepted from asterisk client(1) Fri Aug 4 11:20:17 2006 : disconnected client(2) Fri Aug 4 11:23:02 2006 : accepted from asterisk client(2) Fri Aug 4 11:23:03 2006 : disconnected client(3) Fri Aug 4 11:25:06 2006 : accepted from asterisk client(3) Fri Aug 4 11:25:06 2006 : disconnected server Fri Aug 4 11:28:13 2006 : Festival server started on port 1314 client(1) Fri Aug 4 11:28:23 2006 : accepted from asterisk client(1) Fri Aug 4 11:28:23 2006 : disconnected Here is what shows up in the asterisk logs: Aug 4 11:28:23 DEBUG[16915] chan_sip.c: Setting NAT on RTP to 524288 Aug 4 11:28:23 DEBUG[16915] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 2257: Match Found Aug 4 11:28:23 DEBUG[16915] chan_sip.c: Setting NAT on RTP to 524288 Aug 4 11:28:23 DEBUG[16915] chan_sip.c: Checking SIP call limits for device 101 Aug 4 11:28:23 DEBUG[16915] chan_sip.c: build_route: Contact hop: sip:[EMAIL PROTECTED] Aug 4 11:28:23 VERBOSE[16989] logger.c: -- Executing Answer(SIP/101-081b13a0, ) in new stack Aug 4 11:28:23 DEBUG[16892] channel.c: Avoiding initial deadlock for 'SIP/101-081b13a0' Aug 4 11:28:23 VERBOSE[16989] logger.c: -- Executing Festival(SIP/101-081b13a0, Hello how are you today) in new stack Aug 4 11:28:23 VERBOSE[16989] logger.c: == Parsing '/etc/asterisk/festival.conf': Aug 4 11:28:23 VERBOSE[16989] logger.c: == Parsing '/etc/asterisk/festival.conf': Found Aug 4 11:28:23 DEBUG[16989] app_festival.c: Text passed to festival server : Hello how are you today Aug 4 11:28:23 DEBUG[16989] app_festival.c: Cache file exists, strln=23, strlen=23 Aug 4 11:28:23 DEBUG[16989] app_festival.c: Size OK Aug 4 11:28:23 DEBUG[16989] app_festival.c: Reading from cache... Aug 4 11:28:23 DEBUG[16989] app_festival.c: Passing data to channel... Aug 4 11:28:23 DEBUG[16915] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Response 2258: Match Found Aug 4 11:28:38 DEBUG[16916] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM iax_buddies WHERE ipaddr = '209.33.206.23' AND port = '4569' Aug 4 11:28:38 DEBUG[16916] res_config_mysql.c: MySQL RealTime: Everything is fine. Aug 4 11:29:06 DEBUG[16915] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Aug 4 11:29:28 DEBUG[16916] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM iax_buddies WHERE ipaddr = '209.33.206.23' AND port = '4569' Aug 4 11:29:28 DEBUG[16916] res_config_mysql.c: MySQL RealTime: Everything is fine. The call stays active but nothing is played back from asterisk. Can anyone help me see what I am missing? Thanks in advance for any help. Jon Scottorn Systems Administrator The Possibility Forge, Inc. http://www.possibilityforge.com 435.635.0591 x.1004 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Realtime Macros
Hi, I am trying to get asterisk Realtime to work. I have a fresh installed 1.2.10 setup on a debian system. I have taken the defaul setup and put it into the mysql database. I have setup two extensions 101 and 102. If I setup the extension like such: exten = 101,1,Dial(SIP/101) exten = 102,1,Dial)SIP/102) I can dial back and forth between the two phones. When I switch it to use the stdexten macro and change the extension like such exten = 101,1,Macro(stdexten,101,sip/101) exten = 102,1,Macro(stdexten,102,sip/102) I can not dial each extension and this is what reports on asterisk cli: -- SIP Seeding peer from astdb: '102' at [EMAIL PROTECTED]:5060 for 3600 -- SIP Seeding peer from astdb: '101' at [EMAIL PROTECTED]:1093 for 3600 -- Executing Macro(SIP/101-081a7f90, stdexten,102,sip/102) Jul 24 10:36:37 WARNING[23358]: app_macro.c:149 macro_exec: No such context 'macro-stdexten,102,sip/102' for macro 'stdexten,102,sip/102' == Auto fallthrough, channel 'SIP/101-081a7f90' status is 'UNKNOWN' -- SIP Seeding peer from astdb: '101' at [EMAIL PROTECTED]:1093 for 3600 My question is what has to be in the mysql extenstions_table to get the macro to work? Here is what is in my extensions_table: mysql select * from extensions_table; ++-+---+--+---+--+ | id | context | exten | priority | app | appdata | ++-+---+--+---+--+ | 1 | default | 101 | 1 | Macro | stdexten,101,sip/101 | | 2 | default | 102 | 1 | Macro | stdexten,102,sip/102 | Thanks in advance for any help. Jon Scottorn Systems Administrator The Possibility Forge, Inc. http://www.possibilityforge.com 435.635.0591 x.1004 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] trunk rollover
What kind of line is being used?
in zapata.conf:
group = 1
channel = 1,3,5,6
I create a zap group will all your lines and dial out using the zap group ie...
Dial(Zap/g1/${EXTEN})
By using the group it dials on the first available line.
If you want a more complex setup I have that as well.
I have an agi script that looks at the number dialed and determins if it is a local call if so, dial out the ZAP line, if all ZAP lines are busy dial out an IAX provider, I all IAX lines are busy, then roll to my SIP provider.
Took a bit to figure it all out and get working but it is very useful.
Jon
On Tue, 2006-06-27 at 16:27 -0400, Jim Lynch wrote:
I was hoping that rolling over to the next trunk would be simple, but it
doesn't appear to be so, especially for a newbie. So I'm looking for a
simple way where if I get a busy on the first outgoing trunk, I can do
something to get connected to the next one. Perhaps something like the
big boys do and dial 9 first? I'm guessing a custom dial plan might do
that but I haven't figured out how to do it. I'm running [EMAIL PROTECTED]
version 2.8 sounds right. (maybe that's asterisk 2.8)
Can someone shed some light on a work around until I can figure out
rollovers?
Thanks,
Jim.
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Jon Scottorn
Systems Administrator
The Possibility Forge, Inc.
http://www.possibilityforge.com
435.635.0591 x.1004
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[Asterisk-Users] Grandstream BT101 Auto-Answer
Hi, I am wondering if anyone has gotten the BT101's to work with the paging in Asterisk? I know that the phones themselves have an auto-answer option and if I turn it on every call is auto answered. I want to be able to call the extension normally and have it ring normally but if someone dials # and the extension to have it auto answer for intercom purposes. Anyone have this working? Thanks in advance for any advise or remarks. Jon Scottorn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI ?
On Wed, 2006-05-24 at 00:04 +0200, Freddi Hansen wrote:
From:
Jon Scottorn [EMAIL PROTECTED]
Date:
Tue, 23 May 2006 12:52:02 -0600
To:
Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
On Tue, 2006-05-23 at 19:44 +0100, Thomas Kenyon wrote:
Jon Scottorn wrote:
Hi All,
I have been attempting to get an AGI LCRdialout script to work.
Basically what I need to have happen is when someone dials out a
number the script check to see if it is local if so, go out the ZAP
channel. If the ZAP channel is busy, go out the IAX channels, if IAX
is all busy, go out the SIP channels. Here is a sample of what I have
in my script.
Why can't this be handled directly with the dialplan?
It probably can be but I thought It would be quicker and easier with
AGI. I thought I was supposed to be able to get the variable
DIALSTATUS from asterisk.
Is this not true?
Here are the ways I have been trying but with no success.
$AGI-get_variable(DIALSTATUS);
$AGI-get_variable('DIALSTATUS');
$AGI-get_variable(DIALSTATUS);
$AGI-get_variable(${DIALSTATUS});
try:
my $dialstat = $AGI-get_variable('DIALSTATUS');
Hi,
I have tried that as well.
Thanks for the suggestion.
Any other thoughts.
Thanks,
Jon
Freddi
Any other thoughts anyone might have.
Thanks for the help and input.
*/Jon Scottorn/*
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http://www.possibilityforge.com
435.635.0591 x.1004
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Re: [Asterisk-Users] AGI ?
On Wed, 2006-05-24 at 11:45 -0500, Eric ManxPower Wieling wrote:
my $dialstat = $AGI-get_variable('DIALSTATUS');
Hi,
I have tried that as well.
Thanks for the suggestion.
Any other thoughts.
1) What version of Asterisk are you using?
I am running from debian package version 1.2.7.1.dfsg-2
2) Can you get any other dialplan variables?
I have tried getting other variables and no I do not get any results.
3) Are you running the Dial app inside your AGI or before you run your AGI?
here is what my dialplan looks like within extensions.conf
exten = _5NXX,1,AGI,LCRdialout.agi.test|${EXTEN}
exten = _51NXXNXX,1,AGI,LCRdialout.agi.test|${EXTEN}
Here are the two lines that I run from within the AGI script
$AGI-exec(DIAL Zap/g2/$number|25|TW);
$callStatus = $AGI-get_variable('DIALSTATUS');
$AGI-exec(DIAL IAX2/$iaxUser\:[EMAIL PROTECTED]/$one$areaCode$number|25|TW);
$callStatus = $AGI-get_variable('DIALSTATUS');
Thanks,
Jon Scottorn
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[Asterisk-Users] AGI ?
r\n; return $dialStr; } ### Final catch all for invalid number ### return $dialStr; } Can someone give me some advice as to how to get a channel status of what the $AGI just dialed. Thanks in advance, Jon Scottorn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI ?
I'm sure it could be but taking the agi route I think would be simpler. I thought I should be able to get the variable DIALSTATUS from Asterisk. Is this not true. I have tried these different ways to get it but none have worked yet. $AGI-get_variable(DIALSTATUS); $AGI-get_variable(DIALSTATUS); $AGI-get_variable(DIALSTATUS); On Tue, 2006-05-23 at 19:44 +0100, Thomas Kenyon wrote: Jon Scottorn wrote: Hi All, I have been attempting to get an AGI LCRdialout script to work. Basically what I need to have happen is when someone dials out a number the script check to see if it is local if so, go out the ZAP channel. If the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go out the SIP channels. Here is a sample of what I have in my script. Why can't this be handled directly with the dialplan? Jon Scottorn Systems Administrator The Possibility Forge, Inc. http://www.possibilityforge.com 435.635.0591 x.1004 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI ?
On Tue, 2006-05-23 at 19:44 +0100, Thomas Kenyon wrote:
Jon Scottorn wrote:
Hi All,
I have been attempting to get an AGI LCRdialout script to work.
Basically what I need to have happen is when someone dials out a
number the script check to see if it is local if so, go out the ZAP
channel. If the ZAP channel is busy, go out the IAX channels, if IAX
is all busy, go out the SIP channels. Here is a sample of what I have
in my script.
Why can't this be handled directly with the dialplan?
It probably can be but I thought It would be quicker and easier with AGI. I thought I was supposed to be able to get the variable DIALSTATUS from asterisk.
Is this not true?
Here are the ways I have been trying but with no success.
$AGI-get_variable(DIALSTATUS);
$AGI-get_variable('DIALSTATUS');
$AGI-get_variable(DIALSTATUS);
$AGI-get_variable(${DIALSTATUS});
Any other thoughts anyone might have.
Thanks for the help and input.
Jon Scottorn
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[Asterisk-Users] Auto Dial Out Madness
Hi All, I have been struggling with the auto dial out in asterisk. I am trying to get a call to be auto dialed and play back a message once the line is answered. So far I have been unsuccessful. Currently what happens is I have my .call file. I mv it into /var/spool/asterisk/outgoing. The call is initiated and that all works, my problem is that it does not wait for the line to be answered before playing the message back. It immediately after dialing the number begins playing the message and is done playing it before the person even answers the phone. Does anyone know what I can do to get this working. I want to be able to launch a script from cron that will create the .call file and mv it into the outgoing directory. That is all working currently. Here is a sample of what my .call file looks like and what my extensions.conf looks like. .call Channel: Zap/3/1234567890 Callerid: 1234567890 MaxRetries: 200 RetryTime: 30 WaitTime: 45 Context: outboundmsg1 Extension: s Priority: 1 extensions.conf - snip - [outboundmsg1] exten = s,1,AbsoluteTimeout,40 exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,Answer exten = s,5,Wait(1) exten = s,6,Playback(outboundmsgs/msg1) ; play outbound msg exten = s,7,Background(outboundmsgs/how_to_ack) ; Press 1 to replay or 2 to acknowledge receiving this message exten = 1,1,Goto(s,5) ; replay message exten = 2,1,Goto(msgack,s,1) ; acknowledge message exten = t,1,Playback(vm-goodbye) exten = t,2,Hangup exten = T,1,Hangup Thanks in advance for any suggestions. Jon Scottorn Systems Administrator The Possibility Forge, Inc. http://www.possibilityforge.com 435.635.0591 x.1004 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Upgrade issues
Hi all, I just upgraded asterisk from 1.0.7.dfsg.1-2 to 1.2.7.1.dfsg-2 on a debian system. As I go to restart now, I get this error and can't get asterisk started. [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found May 17 12:11:25 ERROR[13298]: chan_zap.c:7007 mkintf: Unable to get parameters May 17 12:11:25 ERROR[13298]: chan_zap.c:10483 setup_zap: Unable to register channel '1' May 17 12:11:25 WARNING[13298]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1 May 17 12:11:25 WARNING[13298]: loader.c:554 load_modules: Loading module chan_zap.so failed! sato:/etc/asterisk# Ouch ... error while writing audio data: : Broken pipe Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Any help is greatly appreciated. Thanks, Jon Scottorn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
