[Asterisk-Users] Audio problems 50% of the time.
I have an Asterisk server that I use at work. I have a phone that is at home that logs into the Asterisk server at work. My home phone is hooked up via DSL through a Linksys router. You can see the my sip.conf for the phone blow. The problem is each time the phone rings I can hear/be heard 50% of the time. Any suggestion on what to look for. I do have my reg time set for 180 seconds on the cisco ATA186. [72459] type=friend username=XX secret=X host=dynamic context=voice-mail dtmfmode=rfc2833 ;canreivet=yes nat=yes qualify=yes Kurt ___ --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] Audio problems 50% of the time.
I would agree, if I also experience choppy voice. Over the last month I had one spike of 893k over my T1. My average usually is 223k. I carved out 640k for voice QOS on the WAN router. At most I would have 4 calls up at once. The call comes in, the phone rings, 50% of the time I can have a conversations. 50% of time I can not. Maybe I should complain to my SIP service provider. Kurt --- if your connection is also used for web, email, and the worst, p2p, you better to have qos on your router. just be aware that g711 will use 80Kb up and down... gsm and g729 wil use 30/40Kb then : disallow all allow = gsm allow = g729 Olivier kurt x a écrit : I have an Asterisk server that I use at work. I have a phone that is at home that logs into the Asterisk server at work. My home phone is hooked up via DSL through a Linksys router. You can see the my sip.conf for the phone blow. The problem is each time the phone rings I can hear/be heard 50% of the time. Any suggestion on what to look for. I do have my reg time set for 180 seconds on the cisco ATA186. [72459] type=friend username=XX secret=X host=dynamic context=voice-mail dtmfmode=rfc2833 ;canreivet=yes nat=yes qualify=yes Kurt ___ --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] Cisco 2620 as PRI gateway
A 488 can mean a codec miss match. Check that your Asterisk box is configured for g729. Kurt ___ --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] Cisco 2620 as PRI gateway
debug ccsip message Kurt ___ --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] Mulitple voicemail on mulitple phones
I have four DIDs. 2400,2401,2402, and 2403 There is no phone attached to 2400 but the other three DIDs do have phones attached All the four DIDs have their own voicemail and voicemail works on all the DIDs. When you dial 2400 it rings the other three numbers. If no one picks up, it goes to the 2400 voicemail box. What I need to understand is how to notify the other three phones that voicemail was left on the 2400 extension. The other three DIDs must be able to access the 2400 voicemail, and delete it. Any ideas. Kurt ___ --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] Voice recognition
Does anyone know if Asterisk supports any Voice recognition software or is there a third party out that has one available for Asterisk. What I want to do with Voice recognition. When some calls my * IVR instead of the caller spelling the name via the buttons I want the user to be able to say the name. Kurt ___ --Bandwidth and Colocation sponsored by Easynews.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] Having Meetme call another conference
Is it possible to have a bunch of people call a meetme room then have that room call into another conference off net. T Kurt ___ --Bandwidth and Colocation sponsored by Easynews.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] Config PolyCom SoundStation 4000 help
I am trying to get a IP 4000 to register to Asterisk. I can make outbound calls from the IP 4000 but not to it. When I implement sip show peers it lists the extension but with no IP address (unspecified). I am configuring the phone via the web interface. I am not using ftp or tftp to configure the phone. Does anyone have a doc explaining how to get the phone to register to asterisk. Thanks, Kurt ___ --Bandwidth and Colocation sponsored by Easynews.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] asterisk, cisco 3640's and DIDs...
Matching the Correct Inbound POTS Dial Peer for DID For DID to work correctly, make sure the incoming call matches the correct POTS dial-peer where the command direct-inward-dial is configured. If your PRI has DIDs you need the command. Kurt ___ --Bandwidth and Colocation sponsored by Easynews.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] Dial multiple phones
I need to able to ring 30 phones at once on * plus another 10 that are not on Asterisk. I know I can use the Dial(SIP/1SIP/2…SIP/30SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED]/109) but this seems cumbersome. Is there an easier way to do achieve this? Kurt ___ --Bandwidth and Colocation sponsored by Easynews.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] Call Queue ANI
I configured queues.conf and just added a bunch of member = SIP/ numbers to the bottom. I set up my extensions.conf with the access number to the queue. Everything works but the phones on the lists display a ANI of 911 out of area. Is there away to change that ANI to something else. Thanks Kurt ___ --Bandwidth and Colocation sponsored by Easynews.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] Packetization period for CODECs
Is it possible in * to set the Packetization period. For example: If I want G711 to be at 10ms. Is that possible in *? Thanks, Kurt ___ --Bandwidth and Colocation sponsored by Easynews.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] Meetme Problem
I think I configured the MeetMe right. Since I am using SIP for inbound calls I followed the instruction, for 2.6 kernel, from this web page: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy When I call the MeetMe number I get the greeting to enter in your conference room. I do and get invalid conference room. Below is my configs: Executing Wait(SIP/192.168.1.2-08c82740, 1) in new stack -- Executing MeetMe(SIP/192.168.1.2-08c82740, ) in new stack -- Playing 'conf-getconfno' (language 'en') == Parsing '/etc/asterisk/meetme.conf': Found -- Playing 'conf-invalid' (language 'en') Sep 19 10:41:58 WARNING[14066]: file.c:554 ast_readaudio_callback: Failed sip.conf [15551232432] type=friend ;username=2432 ;secret=2432 host=dynamic context=voice-mail dtmfmode=rfc2833 ;canreivet=yes extension.conf [voice-mail] exten = _15551232432,1,wait(1) exten = _15551232432,2,Meetme exten = _15551232432,3,Hangup meetme.conf [voice-mail] conf = 100 Thanks Kurt ___ --Bandwidth and Colocation sponsored by Easynews.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] ztdummy configuration help
Upon setting up and configuring the my extension.conf, meetme.conf and following the instruction outlined at this web page: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy I get the following errors when calling the meetme number. Executing Wait(SIP/216.53.118.2-f41196e0, 1) in new stack -- Executing MeetMe(SIP/216.53.118.2-f41196e0, |sicp) in new stack -- Playing 'conf-getconfno' (language 'en') == Parsing '/etc/asterisk/meetme.conf': Found Sep 19 13:51:22 WARNING[14066]: chan_zap.c:757 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Sep 19 13:51:22 ERROR[14066]: chan_zap.c:6687 chandup: Unable to dup channel: No such file or directory Sep 19 13:51:22 WARNING[14066]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Sep 19 13:51:22 WARNING[14066]: app_meetme.c:230 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') Sep 19 13:51:23 WARNING[14066]: file.c:554 ast_readaudio_callback: Failed to write frame -- Playing 'conf-getconfno' (language 'en') Any help is greatly appreciated. Kurt ___ --Bandwidth and Colocation sponsored by Easynews.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] Polycom Phone advise
I would like to know if any body is using the Polycom Soundstation IP 4000 SIP conference phone with Asterisk. I am thinking of purchasing one. Kurt ___ --Bandwidth and Colocation sponsored by Easynews.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] Comedian annoucment files
A user has their unavailable message played and once that message is over the Comedian message is played right after. Is there any way to prevent the Comedian message being played if the user's unavailable/busy message is being played. Thanks, Kurt ___ 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] Voicemail web access
My problems is when I log into the web page to get my voicemail I see that there nothing being listed. I know there is vmail their because I can retrieve the messages from the phone. I changed the following line in vmail.cgi so I do not need to login with my extension plus context. $context=local; # Define here your by default context (so you dont need to put [EMAIL PROTECTED] in the login I also created a new symbolic link to point to local direct instead to default: lrwxrwxrwx 1 root root 35 Jul 18 11:01 vm - /var/spool/asterisk/voicemail/local Any help would be greatly appreciated. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Timing out issue whenusing AGI
I have the below script that works but for one problem. The call
cannot last longer then 4 minutes when the script is utilized.
However, when I configure my extension.conf to not call the script the
call will stay up until I hang-up.
I call the script as follows:
exten = _24XX,1,AGI(internal.agi|${EXTEN})
exten = _24XX,2,hangup
A brief description of the script is that it allows my asterisk server
to route calls to two different PBXs.
It does not matter which PBX the call is sent to, it will always
hang-up after 4 minutes when using the script.
Any suggestions on what might cause this.
Kurt
#!/usr/bin/perl -w
use warnings;
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-answer(); #I tested with this command pounded and not pounded out.
my $val = $ARGV[0];
open(IN, /var/lib/asterisk/agi-bin/cme_db) or die $!;
my $ext = 0;
print STDERR $val\n;
while (IN) {
chomp;
$ext = $_;
# print STDERR $ext\n;
if ($ext == $val) {
$AGI-exec('Dial',SIP/$ext.'@cme-pbx');
close(IN);
goto EXIT;
}
} # end while loop
close(IN);
open(IN, /var/lib/asterisk/agi-bin/nortel_db) or die $!;
while (IN) {
chomp;
$ext = $_;
# print STDERR $ext\n;
if ($ext == $val) {
$AGI-exec('Dial',SIP/1555123$ext.'@nortel');
close(IN);
goto EXIT;
}
} # end while loop
close(IN);
$rc = $AGI-exec('Dial',SIP/$val.'|15|t');
if ($rc == 0) {
$AGI-exec('Voicemail',u$val);
goto EXIT;
}
EXIT: print STDERR Exiting Script\n;
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[Asterisk-Users] Early media dectection problem
I noticed when I call certain IVR systems, such as 1800calldhl, that Asterisk will not barge the prompt. Would this imply that Asterisk has an Early media detection problem. Is anyone else experiencing this problem. Is there a fix? Kurt ___ 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] Dial peer preference
Does Asterisk support preference for the dial peers. For example: I have two outbound peers in *. The first is a SIP dial peer and the second peer is to the PSTN via a T1. The SIP dial peer is the main outbound peer for all calls. However, if the my SIP providers network goes down, I need to be able to automatically route the call out the T1 card. Is this possible in Asterisk. I have not seen any preference commands for Asterisk. If not, is there a work around for this type of set up. Kurt ___ 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] Multiple Line config help
I have aSIPURA 841 that is working on L1 with phone # 4027. L2 is
configured for 94027.
Both numbers register with Asterisk. When issuing the command sip show peers
both numbers have the same IP address but 94027 show its sip port at
5061. Which I expect is right. When I dial 4027 it works but when I
dial 94027 I get a 486 busy here and voice mail picks up.
config below:
sip.conf
[4027]
type=friend
host=dynamic
dtmfmode=rfc2833
context=home
callerid=SIPURA-L1
canreinvite=no
[EMAIL PROTECTED]
[94027]
type=friend
host=dynamic
dtmfmode=rfc2833
context=home
callerid=SIPURA-L2
canreinvite=no
[EMAIL PROTECTED]
extensions.conf
exten = _40xx,1,Answer
exten = _40xx,2,Dial(SIP/${EXTEN},10,t)
exten = _40xx,3,Voicemail(u${EXTEN})
exten = _40xx,4,Hangup
exten = _40xx,103,Voicemail(b${EXTEN})
exten = _40xx,104,Hangup
exten = _940xx,1,Answer
exten = _940xx,2,Dial(SIP/${EXTEN},10,t)
exten = _940xx,3,Voicemail(u${EXTEN})
exten = _940xx,4,Hangup
exten = _940xx,103,Voicemail(b${EXTEN:1})
exten = _940xx,104,Hangup
Thanks,
Kurt
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Re: [Asterisk-Users] Multiple Line config help
That works. What I am tyring to do is have two separate DIDs. One is
4027 and the
other is 94207. Line 1 = DID 4027 and Line 2 = DID 94027. Dialing
4027 works to line
1 but dial 94027 gets a 486 busy.
Kurt
On 4/21/05, Henry Devito [EMAIL PROTECTED] wrote:
Don't you have to configure your dialplan to hunt to the next extensions?
How else would * know to try 94207 if 4207 is busy?
- Original Message -
From: kurt x [EMAIL PROTECTED]
To: Asterisk asterisk-users@lists.digium.com
Sent: Thursday, April 21, 2005 3:08 PM
Subject: [Asterisk-Users] Multiple Line config help
I have aSIPURA 841 that is working on L1 with phone # 4027. L2 is
configured for 94027.
Both numbers register with Asterisk. When issuing the command sip show
peers
both numbers have the same IP address but 94027 show its sip port at
5061. Which I expect is right. When I dial 4027 it works but when I
dial 94027 I get a 486 busy here and voice mail picks up.
config below:
sip.conf
[4027]
type=friend
host=dynamic
dtmfmode=rfc2833
context=home
callerid=SIPURA-L1
canreinvite=no
[EMAIL PROTECTED]
[94027]
type=friend
host=dynamic
dtmfmode=rfc2833
context=home
callerid=SIPURA-L2
canreinvite=no
[EMAIL PROTECTED]
extensions.conf
exten = _40xx,1,Answer
exten = _40xx,2,Dial(SIP/${EXTEN},10,t)
exten = _40xx,3,Voicemail(u${EXTEN})
exten = _40xx,4,Hangup
exten = _40xx,103,Voicemail(b${EXTEN})
exten = _40xx,104,Hangup
exten = _940xx,1,Answer
exten = _940xx,2,Dial(SIP/${EXTEN},10,t)
exten = _940xx,3,Voicemail(u${EXTEN})
exten = _940xx,4,Hangup
exten = _940xx,103,Voicemail(b${EXTEN:1})
exten = _940xx,104,Hangup
Thanks,
Kurt
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[Asterisk-Users] OutBOund Dial problem
I have the following extension (7700) that can dial out with the below config. exten = _1nxxnxx/7700,1,Dial(SIP/[EMAIL PROTECTED]) exten = _1nxxnxx/7700,2,Hangup If I change it to exten = _1nxxnxx/77XX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _1nxxnxx/77XX,2,Hangup It does not work. Any help is greatly appreciated. Kurt ___ 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] IAXy dial tone problem
I have the Digium S100i IAXy device hooked up to my asterisk server. When I pick up the phone I do get dial tone but it does not stop when I start to dial a number. The dial tone is alway heard and it does not make the call. It does register with Asterisk I can make a call to the IAXy device and here ringing and voice in both direction. I did re-provision the device and reset the device a couple of times. The setup did work yesterday in both directions. The only difference between today and yesterday is the IP address but like I said I did re-provision and reset the device many times. The device is set up for DHCP. Any suggestions. Kurt ___ 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] GotoIf problem
Removing the quotes and eliminating s,3,gotoif did work but its not
what I am looking for.
What I want to do is the following: If a ani that comes in has 10
digits I want to change the ${CALLERIDNUM} to unknown. If the ani
is 10 digits just goto voicemail.
When I set up my [vmail] to look like below, it does not work. When I
send a 4 digit
ani my e-mail confirmation of the voicemail shows the 4 digit ani and
not Unknown.
[globals]
Setvar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM})} != $DIGITS}]?4:5)
exten = s,4,Setvar(${CALLERIDNUM}=Unknown)
exten = s,5,Voicemail(u${ext})
exten = s,6,Hangup
Kurt
On Wed, 09 Mar 2005 07:34:24 +1100, Howard Lowndes [EMAIL PROTECTED] wrote:
On Wed, 2005-03-09 at 05:29, kurt x wrote:
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding is
this venture. When I dial and access voicemail with an ani of 3000
the gotoif statement does not push the call to s|6. Its goes through
each line( 1,2,3,4,5,6,7) . In additon when I dial with a 10 digit
ani the s,3,Gotoif does not work. It also goes through each line(
1,2,3,4,5,6,7)
Any help is greatly appreciated.
Have you tried removing the quotes?
Thanks
Kurt
Asterisk CVS-HEAD-07/14/04-16:28:29 built by
[EMAIL PROTECTED] on a i686 running Linux
[globals]
${ext}=0
SetGlobalVar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM}}) = ${DIGITS}]?s|5)
exten = s,4,GotoIf($[${CALLERIDNUM} = 3000]?s|6)
exten = s,5,Voicemail(u${ext})
exten = s,6,Background(pbx-invalid)
exten = s,7,Hangup
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--
Howard.
LANNet Computing Associates;
Your Linux people http://www.lannetlinux.com
--
When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft.
--
Flatter government, not fatter government;
Get rid of the Australian states.
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Re: [Asterisk-Users] GotoIf problem
To me it looks like the $LEN function is not working. When I do verbose start to * I see that it walks right through every step whether or not the ani is 10 digits or something else. Would it be better to write an AGI script? Kurt On Wed, 09 Mar 2005 11:41:50 -0600, Chris Wade [EMAIL PROTECTED] wrote: kurt x wrote: [globals] Setvar(DIGITS=10) Try this instead... [globals] DIGITS=10 -Chris ___ 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] GotoIf problem
I,ve gotten the GotoIf statement working now. I hard coded the value
10 in place of the ${DIGITS} varible. Worked like a charm.
Thanks to everyone who helped.
Kurt
On Wed, 09 Mar 2005 12:07:51 -0600, Chris Wade [EMAIL PROTECTED] wrote:
kurt x wrote:
[globals]
Setvar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM})} != $DIGITS}]?4:5)
exten = s,4,Setvar(${CALLERIDNUM}=Unknown)
exten = s,5,Voicemail(u${ext})
exten = s,6,Hangup
Oh, and it should be
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM})} != ${DIGITS}]?4:5)
-Chris
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[Asterisk-Users] GotoIf problem
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding is
this venture. When I dial and access voicemail with an ani of 3000
the gotoif statement does not push the call to s|6. Its goes through
each line( 1,2,3,4,5,6,7) . In additon when I dial with a 10 digit
ani the s,3,Gotoif does not work. It also goes through each line(
1,2,3,4,5,6,7)
Any help is greatly appreciated.
Thanks
Kurt
Asterisk CVS-HEAD-07/14/04-16:28:29 built by
[EMAIL PROTECTED] on a i686 running Linux
[globals]
${ext}=0
SetGlobalVar(DIGITS=10)
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM}}) = ${DIGITS}]?s|5)
exten = s,4,GotoIf($[${CALLERIDNUM} = 3000]?s|6)
exten = s,5,Voicemail(u${ext})
exten = s,6,Background(pbx-invalid)
exten = s,7,Hangup
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Re: [Asterisk-Users] GotoIf problem
Will this do.
[general]
static=yes
writeprotect=no
[globals]
${ext}=0
SetGlobalVar(DIGITS=4)
[attendant]
;Main welcome message
exten = xxx2400,1,Answer()
exten = xxx2400,2,Wait(1)
exten = xxx2400,3,Background(welcome)
exten = xxx2400,4,ResponseTimeout,15
;Used for wrong button press
exten = i,1,Goto,invalid|s|1
;To reach the operator
exten = 0,1,Goto,operator|s|1
;Company directory seach feature
exten = 3,1,Directory(local|cme-pbx)
exten = 3,2,Hangup
;To access VoiceMailMain
exten = 9,1,Goto,voicemail|s|1
;Need to be able to dial 4 digit extensions
include = cme-pbx
[voice-mail]
include = attendant
exten = 3000,1,Answer
exten = 3000,2,Dial(SIP/3000)
exten = 3000,3,Hangup
;Used in conjuction with ResposeTimeout
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
;Number that CME dials to forward voice mail to Asterisk
exten = _22XXX,1,Setvar(ext=${EXTEN:1})
exten = _22XXX,2,Goto,vmail|s|1
[vmail]
exten = s,1,Answer
exten = s,2,NoOp(${ext})
exten = s,3,GotoIf($[${LEN(${CALLERIDNUM})} = ${DIGITS}]?s|5)
exten = s,4,GotoIf($[${CALLERIDNUM} = 3000]?s|6)
exten = s,5,Voicemail(u${ext})
exten = s,6,Background(pbx-invalid)
exten = s,7,Hangup
;The below a option triggers on the button
; press *
exten = a,1,VoicemailMain
exten = a,2,Hangup
[cme-pbx]
exten = _24XX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _24XX,2,Hangup
[operator]
exten = s,1,Dial(SIP/[EMAIL PROTECTED])
[voicemail]
exten = s,1,VoicemailMain()
exten = s,2,Hangup
[invalid]
exten = s,1,Playback(pbx-invalid)
exten = s,2,Goto,attendant|xxx2400|3
On Tue, 08 Mar 2005 12:36:38 -0600, Dennis Webb [EMAIL PROTECTED] wrote:
Can you post your dialplan for that extension. Also, NoOp works great for
debugging these issues.
On Tue, 2005-03-08 at 12:29, kurt x wrote:
I am trying to test how the GotoIf and $LEN functions work but am not
succeeding is this venture. When I dial and access voicemail with an ani of
3000 the gotoif statement does not push the call to s|6. Its goes through
each line( 1,2,3,4,5,6,7) . In additon when I dial with a 10 digit ani the
s,3,Gotoif does not work. It also goes through each line( 1,2,3,4,5,6,7) Any
help is greatly appreciated. Thanks Kurt Asterisk CVS-HEAD-07/14/04-16:28:29
built by [EMAIL PROTECTED] on a i686 running Linux [globals]
${ext}=0 SetGlobalVar(DIGITS=10) [vmail] exten = s,1,Answer exten =
s,2,NoOp(${ext}) exten = s,3,GotoIf($[${LEN(${CALLERIDNUM}}) =
${DIGITS}]?s|5) exten = s,4,GotoIf($[${CALLERIDNUM} = 3000]?s|6) exten =
s,5,Voicemail(u${ext}) exten = s,6,Background(pbx-invalid) exten =
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[Asterisk-Users] IAX LAGRQ POKE explanation
Can someone explain in greater detail the following two Control frames. The IAX2 draft document had extremely brief explanations. LAGRQ = Lag request POKE = Poke request. Thanks, Kurt ___ 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] IAX trap question
I would like to know if the following lines represent the RTP traffic going across, the CODEC being used is G711ulaw, or both. The complete trap is below the dotted lines Thanks Kurt asterick*CLI Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 004 Type: VOICE Subclass: 4 Timestamp: 02570ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 02570ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 003 Type: CONTROL Subclass: (255?) Timestamp: 02778ms SCall: 1 DCall: 1 [192.168.2232:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 02778ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 005 ISeqno: 003 Type: VOICE Subclass: 4 Timestamp: 02780ms SCall: 1 DCall: 1 [192.168.2232:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 006 Type: IAX Subclass: ACK Timestamp: 02780ms SCall: 1 DCall: 1 [192.168.2232:4569] -- Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW asterick*CLI Timestamp: 4ms SCall: 1 DCall: 0 [192.168.2232:4569] VERSION : 2 CALLED NUMBER : 2001 CALLING NUMBER : 3000 LANGUAGE: en CALLED CONTEXT : home USERNAME: master asterick*CLI FORMAT : 4 CAPABILITY : 65287 ADSICPE : 2 DATE TIME : 174228915 asterick*CLI Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 9ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI AUTHMETHODS : 3 CHALLENGE : 759448742 USERNAME: master asterick*CLI Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 9ms SCall: 1 DCall: 1 [192.168.2232:4569] MD5 RESULT : 707018f7eb07cfa8a966853c868683a7 asterick*CLI Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACCEPT Timestamp: 00012ms SCall: 1 DCall: 1 [192.168.2232:4569] FORMAT : 4 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00012ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass: ANSWER Timestamp: 00015ms SCall: 1 DCall: 1 [192.168.2232:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 00015ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Mar 2 16:45:38 NOTICE[-1222640720]: rtp.c:285 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible asterick*CLI Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 002 Type: CONTROL Subclass: RINGING Timestamp: 00018ms SCall: 1 DCall: 1 [192.168.2232:4569] Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 00018ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 004 Type: VOICE Subclass: 4 Timestamp: 02570ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 003 Type: IAX Subclass: ACK Timestamp: 02570ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 003 Type: CONTROL Subclass: (255?) Timestamp: 02778ms SCall: 1 DCall: 1 [192.168.2232:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 02778ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 005 ISeqno: 003 Type: VOICE Subclass: 4 Timestamp: 02780ms SCall: 1 DCall: 1 [192.168.2232:4569] Tx-Frame Retry[-01] -- OSeqno: 003 ISeqno: 006 Type: IAX Subclass: ACK Timestamp: 02780ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 006 Type: IAX Subclass: HANGUP Timestamp: 08033ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI Rx-Frame Retry[No] -- OSeqno: 006 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 08033ms SCall: 1 DCall: 1 [192.168.2232:4569] asterick*CLI ___ 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] IAX channel unable to create
I have two * boxes running two differnet versions of *. Box A is running: Asterisk CVS-HEAD-07/14/04-16:28:29 built by [EMAIL PROTECTED] on a i686 running Linux Box B is running: Asterisk 1.0.3 built by [EMAIL PROTECTED] on a i386 running FreeBSD I can make a IAX call from B to A but not from A to B. When I try to make a call from A to B I get these messages: Feb 21 12:48:12 WARNING[-1233155152]: channel.c:1860 ast_request: No channel type registered for 'IAX' Feb 21 12:48:12 NOTICE[-1233155152]: app_dial.c:696 dial_exec: Unable to create channel of type 'IAX' Feb 21 12:48:14 WARNING[-1116300368]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 1 (Non-critical Response) My box A iax.conf: [general] port=5036 bindport=5036 bandwidth=low allow=ulaw disallow=lpc10 jitterbuffer=no tos=lowdelay [slave] type=friend secret=4435 context=voice-mail defaultip=192.168.2.232 qualify=yes My Box A extension.conf [voice-mail] exten = _2000,1,Dial(IAX/master:[EMAIL PROTECTED]/[EMAIL PROTECTED]) My box B iax.conf [general] port=5036 bindport=5036 bandwidth=low allow=ulaw disallow=lpc10 tos=lowdelay [master] type=friend secret=4435 context=home defaultip=192.168.1.2 qualify=yes My Box B extension.conf [home] exten = _24xx,1,Dial(IAX2/slave:[EMAIL PROTECTED]/[EMAIL PROTECTED]) Thanks in advance Kurt ___ 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] VoiceMail ANI question
When I receive voicemail notification via e-mail I noticed that the
${VM_CALLERID) puts the IP address of the * box when callee info is
not present. Is there a way to have the field put Unkown caller in
instead of the IP address of the * box.
Kurt
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[Asterisk-Users] Rining Issues
When I access the Directory() and use it to call an extension, the origination hears garbled or inconsistent ringing. The termination side rings normally and the conversation is clean in both directions. Kurt ___ 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] Directory() ringing problem
The Directory command is working properly but the ringing herd in the origination phone is either garbled or herd infrequently. The termination phone does ring with consistency. Any suggestion on what might be happening. Kurt ___ 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] IVR Timing out
I set up an IVR systems that plays a message for 15 seconds but once the message is over you can not select any of the prompts. If you select something within 10 seconds the IVR system works. I even set the ResponseTimeout cmd to 25 secs but that does not work. Jan 24 09:54:29 NOTICE[-1222644816]: sched.c:221 sched_settime: Request to schedule in the past?!?! [attendant] ;Main welcome message exten = s,1,Wait(2) exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,25 exten = s,4,Background(welcome_n2p1) exten = s,5,Hangup Thanks in advance for help, Kurt ___ 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] IVR Timing out
Did what you suggested but with the same results plus the following error messages: [attendant] ;Main welcome message exten = s,1,Wait(2) exten = s,2,DigitTimeout,5 exten = s,3,AbsoluteTimeout(25) exten = s,4,Background(welcome_n2p1) ;exten = s,4,ResponseTimeout,25 ;exten = s,5,Hangup exten = s,5,Background(silence/10) exten = s,6,Background(silence/10) exten = s,7,Goto(s,5) Jan 24 13:10:51 WARNING[-1233134672]: file.c:473 ast_openstream: File silence/10 does not exist in any format Jan 24 13:10:51 WARNING[-1233134672]: file.c:761 ast_streamfile: Unable to open silence/10 (format ULAW): No such file or directory Kurt On Mon, 24 Jan 2005 12:53:11 -0500, Roger Gulbranson [EMAIL PROTECTED] wrote: On Mon, 2005-01-24 at 12:36 -0500, kurt x wrote: . Once the .gsm file is finished playing you can not select any of the menu items. The .gsm file is roughly 15 to 17 seconds long.If you make a selection before the .gsm file finishing playing you can select any of the menu items. OK, then following my previous advice, alter your s extension to: exten = s,1,Wait(2) exten = s,2,DigitTimeout(5) exten = s,3,AbsoluteTimeout(25) exten = s,4,Background(welcome_n2p1) exten = s,5,Background(silence/10) exten = s,6,Background(silence/10) exten = s,7,GoTo(s,5) ___ 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] Voicemail.conf pin protection
Is there any way to encrypt the PIN numbers in voicemail.conf. I looked at the Wiki page for voicemail.conf but it did not mention anything about that topic. I am not using MySQL or any other thrid party database. Kurt ___ 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] SIP debugs
Other then the standard sip debug is there any other sip debug bugs like for errors, events, etc. Kurt ___ 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] Accessing Voice mail
Brain,
I did what you suggested but instead of going to VoiceMailMain it
starts the begining of
my recorded message each time I press the * key.
[vmail]
exten = a,1,Voicemail(u${ext})
exten = a,2,Hangup
Kurt
On Wed, 19 Jan 2005 11:48:18 -0500, Brian Dingman [EMAIL PROTECTED] wrote:
If you put the following in your Dialplan, pressing * should break you
out of voicemail and call VoiceMailMain
exten = a,1,VoicemailMain,EXTEN
exten = a,2,Hangup
On Wed, 19 Jan 2005 11:33:23 -0500, kurt x [EMAIL PROTECTED] wrote:
I want to know if there is way to break out of the voicemail message.
for example:
On my Noterl PBX when you dial you number from any where
you get your recorded voice mail message, but during the message I
press 81 and break out of that message. It then
prompts me for my PIN thus allowing me to access my message
without using the auto attendant.
Is this possible with Comedian?
The below page did help.
http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain
Kurt
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[Asterisk-Users] Accessing Voice mail
I want to know if there is way to break out of the voicemail message. for example: On my Noterl PBX when you dial you number from any where you get your recorded voice mail message, but during the message I press 81 and break out of that message. It then prompts me for my PIN thus allowing me to access my message without using the auto attendant. Is this possible with Comedian? The below page did help. http://www.voip-info.org/wiki-Asterisk+cmd+VoiceMailMain Kurt ___ 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] Outbound Dial via SIP
What I am trying to do is the following: A call is sent to the * box
via a SIP invite. The * box answers via an IVR menu system with
Enter the extension you want to dial so I enter in my 5 digit
extension and get the below message.
Jan 18 10:10:03 WARNING[-1380238416]: channel.c:1860 ast_request: No
channel type registered for 'SIP)'
Jan 18 10:10:03 NOTICE[-1380238416]: app_dial.c:696 dial_exec: Unable
to create channel of type 'SIP)'
Jan 18 10:10:05 WARNING[-1115923536]: chan_sip.c:673 retrans_pkt:
Maximum retries exceeded on call
[EMAIL PROTECTED]
for seqno 1 (Non-critical Response)
My extension.conf outbound dial peer:
[outbound]
exten = _124XX,1,Dial(SIP)/${EXTEN:[EMAIL PROTECTED])
exten = _124XX,2,Playback(invalid)
exten = _124XX,3,Hangup
My sip.conf
[outbound]
type=peer
host=192.168.1.1
What the * needs to do is receive the call via SIP and then send it
out dialed extension via SIP to an another IP PBX. SO the * does not
need to register to a server just blindly send a SIP invite to the ip
address in the SIP.CONF file: 192.168.1.1
Any help would be appricated
Kurt
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Re: [Asterisk-Users] Outbound Dial via SIP
That was the ticket. The Extra ) was the problem.
Thanks Sean.
Kurt
On Tue, 18 Jan 2005 08:13:31 -0800, Sean Kennedy [EMAIL PROTECTED] wrote:
kurt x wrote:
What I am trying to do is the following: A call is sent to the * box
via a SIP invite. The * box answers via an IVR menu system with
Enter the extension you want to dial so I enter in my 5 digit
extension and get the below message.
Jan 18 10:10:03 WARNING[-1380238416]: channel.c:1860 ast_request: No
channel type registered for 'SIP)'
Jan 18 10:10:03 NOTICE[-1380238416]: app_dial.c:696 dial_exec: Unable
to create channel of type 'SIP)'
Jan 18 10:10:05 WARNING[-1115923536]: chan_sip.c:673 retrans_pkt:
Maximum retries exceeded on call
[EMAIL PROTECTED]
for seqno 1 (Non-critical Response)
My extension.conf outbound dial peer:
[outbound]
exten = _124XX,1,Dial(SIP)/${EXTEN:[EMAIL PROTECTED])
exten = _124XX,2,Playback(invalid)
exten = _124XX,3,Hangup
My sip.conf
[outbound]
type=peer
host=192.168.1.1
What the * needs to do is receive the call via SIP and then send it
out dialed extension via SIP to an another IP PBX. SO the * does not
need to register to a server just blindly send a SIP invite to the ip
address in the SIP.CONF file: 192.168.1.1
Any help would be appricated
Kurt
exten = _124XX,1,Dial(SIP)/${EXTEN:[EMAIL PROTECTED])
This is your problem. It should be
exten = _124XX,1,Dial(SIP/${EXTEN}:[EMAIL PROTECTED])
There might be further syntax errors, this is only off the top of my head,
but the most glaring error that I could see was the ) after SIP.
Sean
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[Asterisk-Users] Directory() Command
I am trying to use the Directory() but am having difficulty using it. According to Wiki page that I found you need to pass it your voicemail.conf context. My vm-context is [local]. So when I setup the cmd (Directory(local)) I can search on the three letters of the last name find that user. But once I press one to except the name and dial the extension I get the following message form the * CLI. Jan 17 15:22:07 WARNING[-1285669968]: app_directory.c:182 play_mailbox_owner: Can't find extension '' in context 'local'. Did you pass the wrong context to Directory? Reading the above error message I see that I need to pass it my outbound context. So I setup the command to look as follows: Directory(local outbound). I reload * and try again but this time it does not even pick up the name I search for. I used the same name in the first example. Any ideas on where I want wrong would be greatly appreciated. Kurt ___ 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] RE: Cisco Unity and Asterisk
Question: What is your reasoning for using Cisco Voice Mail instead of Asterisk's voice mail. IMHO it would make more sense to keep everything on Asterisk. Kurt ___ 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] RE: DTMF tones from CCME phone
You need to either download 12.3(11)T or 12.3(10)LD. Kurt ___ 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] RE: Cisco to * problem
See if you have the below configure under your dial peers or voice service voip. If you do, then issue this command no signaling forward unconditional signaling forward unconditional Kurt ___ 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] Asterisk PBX and backup Circuits
I am interested to know how one would calculate the amount of PSTN connection needed for backup on an Asterisk PBX that is being setup to receive its DIDs via a VoIP provide. To sum up what I am implementing: I am porting my DIDs to a VoIP provide so I will need a back up plan in place if the Data network fails. In addition, 911 will always be going out the PSTN so I know I need at least one POTs circuit. Calls inbound and outbound will always routed through the data network. Thanks, Kurt ___ 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
