Re: [asterisk-users] Call Max Time
On 08/27/06 13:23 Rushowr said the following: Set(TIMEOUT(absolute)=seconds) Change seconds to the number of seconds you want to allow a call to last alternatively, look at the L() option to Dial. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Nobody is responding. Why? (Implement music on transfer)
On 08/26/06 23:52 Crazy Boy said the following: Hi friends, I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, i'm not exactly sure what you're intending to do, but MoH is already active and played for attended transfers. blind transfers will relay the call indication tones. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call Max Time
Hello,Could you tell how i can use it in PERL AGI script?currently i am using in my AGI with this format, but some time call is not disconnecting customers talking without money.$dialstr = "SIP/terminator/15745405022|350|tTL(653044:7000:5000)";$AGI-exec('Dial', $dialstr);regards,
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RE: [asterisk-users] Call Max Time
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
First big question is are you checking beforehand how long the limit should
be by calculating ((BALANCE / RATE) / 1000)
If you're not, that would be why it doesn't disconnect the customer within a
time period that wouldn't result in a negative balance.
Other than that, you might want to possibly check if your script is getting
the dialstring properly. Do you need to escape the / characters in it? What
I'd personally do is set up some Verbose() statements in my scripts to
output debugging data.
Hope this helps!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abdul
Sent: Sunday, August 27, 2006 2:54 AM
To: Asterisk-Users@lists.digium.com
Subject: RE: [asterisk-users] Call Max Time
Hello,
Could you tell how i can use it in PERL AGI script?
currently i am using in my AGI with this format, but some time call
is not disconnecting customers talking without money.
$dialstr = SIP/terminator/15745405022|350|tTL(653044:7000:5000);
$AGI-exec('Dial', $dialstr);
regards,
Get your own web address for just $1.99/1st yr
http://us.rd.yahoo.com/evt=43290/*http://smallbusiness.yahoo.com/domains .
We'll help. Yahoo! Small Business
http://us.rd.yahoo.com/evt=41244/*http://smallbusiness.yahoo.com/ .
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Comment: ENCRYPTED WITH GPG
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WclJ5IBoFFF1NBdDb3P/oXM=
=9Jxz
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First big question is are you checking beforehand how long
the limit should be by calculating ((BALANCE / RATE)/
1000)
If you're not, that would be why it doesn't disconnect the
customer within a time period that wouldn't result in a negative balance.
Other than that, you might want to possibly check if your
script is getting the dialstring properly. Do you need to escape the /
characters in it? What I'd personally do is set up some Verbose() statements in
my scripts to output debugging data.
Hope this helps!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
AbdulSent: Sunday, August 27, 2006 2:54 AMTo:
Asterisk-Users@lists.digium.comSubject: RE: [asterisk-users] Call
Max Time
Hello,Could you tell how i can use it in PERL AGI
script?currently i am using in my AGI with this format, but some time
call is not disconnecting customers talking without money.$dialstr =
"SIP/terminator/15745405022|350|tTL(653044:7000:5000)";$AGI-exec('Dial',
$dialstr);regards,
Get your own web
address for just $1.99/1st yr. We'll help. Yahoo!
Small Business.
PGPexch.htm.asc
Description: PGP signature
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Re: [asterisk-users] IP phone with 2 ethernet jacks
Mario wrote: We have used both IP501, IP601 (Polycom), Snom 320 and Snom 360. All of them are good phones with very good quality of voice and full of features. However, SNOM phones have a feature (missing from Polycom) that most of our customers really require: with SNOM phones you have leds for presence support that allow you to see which other extensions are busy (through the Asterisk Hint command). If this is important for you, you should really stay with Snom. This feature isn't missing from Polycom phones, although it may not work the same as what you want. But the phones do support the presense feature, and I have it working on my IP501's with Asterisk. John ___ --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] Call Max Time
Hi,i am using the same calculating ((BALANCE / RATE) / 1000) method to return tTL.and i am sure my GAI is working well. but could u tell me how i can set Verbose() sepecial for my dialstring?Regards, Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --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] Call Max Time
from within asterisk, just run the following command: show application Verbose That'll fill you in. Your other solid option is to search the wiki From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AbdulSent: Sunday, August 27, 2006 4:05 AMTo: Asterisk-Users@lists.digium.comSubject: RE: [asterisk-users] Call Max Time Hi,i am using the same calculating ((BALANCE / RATE) / 1000) method to return tTL.and i am sure my GAI is working well. but could u tell me how i can set Verbose() sepecial for my dialstring?Regards, Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --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] Voicemail's mail formate
Dear all I am using AMP 1.0.10 . my voicemail system working perfectly, but now i like to sent user PIN ( Password of the extension) number with that mail . how could i read the user passwd value (PIN) so that i could append the mail format thanks Salaque ___ --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] Re: Wellgate 3804a
On 2006-08-24 08:43:01 -0700, Ronald Wiplinger [EMAIL PROTECTED] said: I want that each call from PSTN goes to Asterisk to the context for this line. Within this context can be a menu or a dial command, ... As more I read, as more I get confused, ... and each try is not working! I don't know if it's the same, but I am using the wellgate 3701A (1FXO/FXS) box with asterisk and it's working all right. My sip.conf: [WG88621001] type=friend defaultip=192.168.250.244 insecure=very context=incoming_WG dtmfmode=rfc2833 [EMAIL PROTECTED] language=en nat=yes auth=md5 host=dynamic canreinvite=no disallow=all allow=ulaw allow=alaw allow=g726 allow=g729 username=88621001fromuser=88621001 secret= qualify=yes canreinvite=no extensions.conf [incoming_WG] exten = s,1,NoOp(*** I am here now ***) Wellgate settings: Network Interface: IP address of the device 192.168.250.244 Sip Config: Mode Proxy Primary Proxy IP address: 192.168.250.20 Line 1 Number:88621001 Security Config Line1 Account: WG8862001 Line 1 password: (secret from the asterisk setting) Line configuration Line 1 (LINE) Type: FXO Hunting Group: 1 HotLine: 601 Registration: Not Registered Status: Ready System Configuration Keypad type: rfc2833 Routing Table Index: IP Default Destination: FXO E.164: x Index: FXO Destination: IP Default E.164: x *CLI sip show peers like ^WG Name/username HostDyn Nat ACL Port Status WG88621001/88621001(Unspecified)D N 0 UNKNOWN 1 sip peers [0 online , 1 offline] Calls from PSTN comes to the IVR asking for the extension number and than nothing happens. Asterisk shows nothing either. Can somebody enlighten me: 1. Do I need to have a register statement in sip.conf? (I tried register = 88621001:secret-from-above ; Wellgate GW.3801-Line-1) 2. where to turn off the IVR? 3. Do I use the right name, user name, line account, line 4. Hotline. Why, how, which number?? I have to sleep now, but I will review the setting on my rig tomorrow. I seem to remember the routing table was the key to making it work... Till tomorrow. Marty ___ --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] H323
any one try that with g723 codec? thanks Salaque On 8/27/06, Rosli Sukri [EMAIL PROTECTED] wrote: i am also using ooh323 - it works fine on sjphone ekiga etc but i cant seem to get it to work with ms netmeeting On 8/26/06, atik khan [EMAIL PROTECTED] wrote: Hi, i used to work ooh323 with my asterisk. it gives better performance than other oh323 or H323 comes with asterisk... i got H323 channel and oh323 with a lot of error.( like codec selection )but ooh323 works fine with me thanks atik On 26 Aug 2006 12:13:52 +0200, andrutto [EMAIL PROTECTED] wrote: Hi What is the best solution for H323 in asterisk -- h323 in source, -- oh323 or -- ooh323c? which is most robust and reliable? Which supports gatekeeper functionality? Best wishes Andrutto -- Najnowsze fakty!!! http://link.interia.pl/f1996 ___ --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 ___ --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 ___ --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 -- Got my Gmail ? ___ --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] about MusicOnHold / Playback
Hi, I have a problem about musiconhold/playback command in the dial plan. Actually, I have the following dial plan for inbound call. [from-gateway] exten = 1234,1,Answer() exten = 1234,2,MusicOnHold(c1) exten = 1234,3,Hangup() When I using mobile to make call 1234, it works and I can hear music. (below is the log from CLI) -- Executing Answer(SIP/203.193.26.242-087233a8, ) -- Executing MusicOnHold(SIP/203.193.26.242-087233a8, ) -- Started music on hold, class 'c1', on channel 'SIP/203.193.26.242-087233a8' Aug 27 15:04:36 NOTICE[18840]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 203.193.26.242 Then, I use a normal phone to make a call to 1234 thro' PSTN, I can't hear any music and CLI shows there is a looping. In the log below, it tries to play the music but somehow it stops to play and go back to answer to call again. It shows spawn extension (from-gateway,1234,2) exited It seems that there is a problem in this command. However, I can't find any hint in the message and even log. -- Executing Answer(SIP/203.193.26.242-08723500, ) in new stack -- Executing MusicOnHold(SIP/203.193.26.242-08723500, c1) in new stack -- Started music on hold, class 'c1', on channel 'SIP/203.193.26.242-08723500' -- Stopped music on hold on SIP/203.193.26.242-08723500 == Spawn extension (from-gateway, 1234, 2) exited non-zero on 'SIP/203.193.26.242-08723500' -- Executing Answer(SIP/203.193.26.242-08728a10, ) in new stack -- Executing MusicOnHold(SIP/203.193.26.242-08728a10, c1) in new stack -- Started music on hold, class 'c1', on channel 'SIP/203.193.26.242-08728a10' -- Stopped music on hold on SIP/203.193.26.242-08728a10 == Spawn extension (from-gateway, 1234, 2) exited non-zero on 'SIP/203.193.26.242-08728a10' -- Executing Answer(SIP/203.193.26.242-08723500, ) in new stack -- Executing MusicOnHold(SIP/203.193.26.242-08723500, c1) in new stack -- Started music on hold, class 'c1', on channel 'SIP/203.193.26.242-08723500' -- Stopped music on hold on SIP/203.193.26.242-08723500 == Spawn extension (from-gateway,1234, 2) exited non-zero on 'SIP/203.193.26.242-08723500' Does anyone have the same problem as me? Anyone can suggest me a solution to solve it? Thanks. ___ --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] Doubled digits on vm pasword
Hi all, I'm running Asterisk SVN-trunk-r40489 on which one I have a Sipura 1001 connected. I face a problem when sending digits to voicemail password: each one is sended twice (eg 35 give 3355) I have the same behaviour if I have to enter the mailbox number before. I have no problem to navigate in voicemail options or so. DTMF is set to auto, I also tested dtmf2833 and inband. A software client like KIAX has not the problem. I called through PSTN a IVR number, working like a charm! If someone had any idea on what could be the problem, thanks in advance. -- Daniel ___ --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] Re: SV: E61
On 27 Aug 2006, at 04:03, Dovid Bender wrote: I was not going to get it based on what people said about the E61 and the NAT issues. Is this false ? I was thinking of getting it for when I travel to Israel. There seems to be a lot of open wifi connections all over the country there. Also how is the radio for the wifi on it ? Speaking for the E60, the Wifi radio is kind of shitty. It likes to just disconnect out of the blue and will only reconnect if the phone is rebooted. When the connection is up you can get decent download speeds and SIP calls are crystal clear, though ;) I never tried to use the SIP client through NAT. My Asterisk is on the same LAN. jens ___ --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] Dial C option
Hello
I would like to NOT record a CDR for internal calls, but the C option
(suppose to work like NoCDR() ) is just not working for me. My dial line is
exten = _70XX,1,Dial(SIP/${EXTEN}|20|Ctr)
Could someone give me a short example of using NoCDR correctly.
Thanks
Master
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[asterisk-users] asterisk registering as extension to another asterisk server problem
Hi. If I register asterisk with another server as an extension to that server -- say -- using iax2 how can I dial an extension on that second server? I tried the following exten = 8200,1,Dial(iax2/201/8200,,r) but got no route to destination even though the other server saw my registration. What am I missing here? Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ --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] can not get ${LEN(VAR)} and greater than to work for me
On Saturday August 26 2006 11:15 am, Matt Riddell (IT) wrote:
John Millican wrote:
Hello all,
I am trying to test if the length of a dialed number is greater than 7.
When i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}7]?dialout:nodial);
and I dial an 11 digit number i.e. 1 800 xxx
i get this in the console:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 0?dialout:nodial) in
new stack
indicating that the number was not greater than 7.
if i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}=11]?dialout:nodial);
and dial the same 1 800 xxx
i get:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 1?dialout:nodial) in
new stack
indicating that the length of number dialed was equal to 11 digits.
so equal to works and greater than does not?
Can any one see what I am doing wrong?
* version 1.2.9.1
Maybe string comparison because of the speech marks?
Thank You Matt and Ira
The speech marks/quotes were the problem.
Matt sorry about the earlier direct mail used R instead of L for the reply.
John M
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Re: [asterisk-users] IP phone with 2 ethernet jacks
You can get that on the polycom if you want to fork over another $200.00 + for the side car. Or if you are using a 601 you can use the first line for all your calls and then the next 5 for it. - Original Message - From: Mario [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 26, 2006 5:08 AM Subject: Re: [asterisk-users] IP phone with 2 ethernet jacks We have used both IP501, IP601 (Polycom), Snom 320 and Snom 360. All of them are good phones with very good quality of voice and full of features. However, SNOM phones have a feature (missing from Polycom) that most of our customers really require: with SNOM phones you have leds for presence support that allow you to see which other extensions are busy (through the Asterisk Hint command). If this is important for you, you should really stay with Snom. Guido Hecken wrote: We like the SNOM 360 Phones. They have really good features. Guido -Ursprüngliche Nachricht- Von: Mindaugas Kuprys [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 25. August 2006 09:40 An: asterisk-users Betreff: [asterisk-users] IP phone with 2 ethernet jacks Hi, Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wanted Sipura but they don't have such product. Thanks, Mindaugas ___ --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 ___ --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 ___ --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] Re: SV: E61
On 27 Aug 2006, at 04:03, Dovid Bender wrote: I was not going to get it based on what people said about the E61 and the NAT issues. Is this false ? I was thinking of getting it for when I travel to Israel. There seems to be a lot of open wifi connections all over the country there. Also how is the radio for the wifi on it ? Speaking for the E60, the Wifi radio is kind of shitty. It likes to just disconnect out of the blue and will only reconnect if the phone is rebooted. When the connection is up you can get decent download speeds and SIP calls are crystal clear, though ;) I never tried to use the SIP client through NAT. My Asterisk is on the same LAN. jens Anyone know if the E61 is any diffrent ? ___ --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] Annoying Bristuff
HiI change the kernel to 2.6.14.7, but unfortunately the problem still exist.The messages empty HDLC frame or bad CRC received appear only when there is not traffic on card (0 active calls). It never happens during a call. Strange?!? Any other tips are gladly expected :).CheersAndrew ___ --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] Can the codec/format for name/greeting in voicemail be changed?
- RR [EMAIL PROTECTED] wrote: Any ideas if it's possible to either record greetings/names in a different format than GSM OR be able to convert these voicemail subscriber greetings in my database to some other format? They will be recorded in the same formats that you record voicemail messages in, which you can control via settings in voicemail.conf. Converting existing recordings can be done with 'sox' (except for G.729, of course). -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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] Re: SV: E61
NAT is a problem at the moment, I can only connect to my Asterisk server on the same network. Wifi work nicely and you can get up groups of access points so that when you move it roams to the next active point you're on. I hear Nokia are aware of the NAT issue and are going to update.RegardsRobOn 27/08/06, Dovid Bender [EMAIL PROTECTED] wrote: On 27 Aug 2006, at 04:03, Dovid Bender wrote: I was not going to get it based on what people said about the E61 and the NAT issues. Is this false ? I was thinking of getting it for when I travel to Israel. There seems to be a lot of open wifi connections all over the country there. Also how is the radio for the wifi on it ? Speaking for the E60, the Wifi radio is kind of shitty. It likes to just disconnect out of the blue and will only reconnect if the phone is rebooted. When the connection is up you can get decent download speeds and SIP calls are crystal clear, though ;) I never tried to use the SIP client through NAT. My Asterisk is on the same LAN. jensAnyone know if the E61 is any diffrent ?___--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 ___ --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] [RESOLVED] One way audion on Sangoma
When I do echocancel=yes it stops working. I have to have it at no in order for it to work. - Original Message - From: Dovid Bender [EMAIL PROTECTED] To: Dovid Bender [EMAIL PROTECTED] Sent: Friday, August 25, 2006 12:23 PM Subject: Fw: [asterisk-users] [RESOLVED] One way audion on Sangoma - Original Message - From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, August 25, 2006 11:43 AM Subject: Re: [asterisk-users] [RESOLVED] One way audion on Sangoma On 11:24, Fri 25 Aug 06, Andrew Kohlsmith wrote: Digium hardware echo cancellation cards also require you to say echocancel=yes in zapata.conf. The zaptel driver recognizes that the card possesses echo cancellation hardware and does not engage the software echo canceller for those channels. To summarize: if you want echo cancellation on Zaptel channels, you must enable it (echocancel=yes, or a number of taps) in zapata.conf. If hardware echo cancellation exists, it is used over software echo cancellation. Note that if hardware echo cancellation hardware is detected, the # of taps is ignored and the hardware uses whatever it has internally. echocancel=no/off in zapata.conf will disable the echo cancellation in Zaptel, whether it is hardware-based or software-based. thnx, I'll now go back to reading docs before I say anything stupid again. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --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 ___ --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] CDR Function - Asterisk-1.2.10
Hi,
I'm having problems with calling the ${CDR(billsec)} ${CDR(duration)}
variables in an AGI.
Note that I'm using Asterisk-1.2.10 and Realtime extensions + Realtime sip
users/peers.
John
mail2web - Check your email from the web at
http://mail2web.com/ .
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Re: [asterisk-users] CDR Function - Asterisk-1.2.10
Can you give us some more info? Like agi debug output?
On 8/27/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I'm having problems with calling the ${CDR(billsec)} ${CDR(duration)}
variables in an AGI.
Note that I'm using Asterisk-1.2.10 and Realtime extensions + Realtime sip
users/peers.
John
mail2web - Check your email from the web at
http://mail2web.com/ .
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[asterisk-users] Cannot dial out through SIP provider
Hi,
I'm running Asterisk 1.2.10 bristuffed.
Asterisk is registring perfectly against my provider (musimi.dk), and
incoming calls comes in and are routed fine to either internal ZAP
(ISDN BRI) and/or SIP.
But
I can't dial out via SIP (musimi)
sip.conf:
[musimi]
type=friend
host=musimi.dk
username=
fromuser=
secret=xx
domain=musimi.dk
fromdomain=musimi.dk
context=from-sip
;nat=yes
;canreinvite=no
insecure=very
dtmfmode=rfc2833
[]
type=friend
context=internal
username=
secret=
host=dynamic
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=ulaw
callerid=Henrik Woffinden
nat=yes
qualify=yes
insecure=very
;[EMAIL PROTECTED]
extensions.conf:
[internal]
;exten = _,1,Dial(Zap/g1/${EXTEN},,)
exten = _,1,Dial(SIP/[EMAIL PROTECTED],,)
exten = _,n,Hangup
If I want to dial out via ISDN (Zap which is commented out above), then
it works ok, but via SIP I get the following error message (my own
number is and the number I dial is - which is a normal
mobile):
-- Registered SIP '' at 192.168.9.9 port 29796 expires 3600
-- Executing Dial(SIP/-09f2eb28, SIP/[EMAIL PROTECTED]||) in new stack
-- Called [EMAIL PROTECTED]
Aug 27 16:38:41 NOTICE[23387]: chan_sip.c:9845 handle_response_invite:
Failed to authenticate on INVITE to 'Henrik Woffinden
sip:[EMAIL PROTECTED];tag=as06ed5480'
-- SIP/musimi-09f34188 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup(SIP/-09f2eb28, ) in new stack
== Spawn extension (internal, , 2) exited non-zero on
'SIP/-09f2eb28'
I hope somebody can tell me what I'm doing wrong here.
--
Med venlig hilsen / Best regards,
Henrik Woffinden
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Re: [asterisk-users] CDR Function - Asterisk-1.2.10
AGI
===
$res = $AGI-exec(Hangup);
$foo = ${CDR(billsec)};
myVerbose($foo); #print on CLI
$foo = ${CDR(duration)};
myVerbose($foo);
$foo = ${CDR(answer)};
myVerbose($foo);
$foo = ${CDR(start)};
myVerbose($foo);
when exected with perl
Undefined subroutine main::CDR
Im passing ${EXTEN} ${CALLERIDNUM} variables through extension table. agi
accepts them fine.
Original Message:
-
From: Justin Tunney [EMAIL PROTECTED]
Date: Sun, 27 Aug 2006 11:41:02 -0400
To: [EMAIL PROTECTED], asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CDR Function - Asterisk-1.2.10
Can you give us some more info? Like agi debug output?
On 8/27/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I'm having problems with calling the ${CDR(billsec)} ${CDR(duration)}
variables in an AGI.
Note that I'm using Asterisk-1.2.10 and Realtime extensions + Realtime sip
users/peers.
John
mail2web - Check your email from the web at
http://mail2web.com/ .
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[asterisk-users] detecting a users number using the dialplan or AGI
Hi All,I was hoping someone could help me with a problem I'm having determining a users number. Is there any way in the dialplan or with an AGI to detect what a users number is for use in a meetme conference? I am using the MeetMeAdmin function from within the dialplan.I would like one of my admins to be able to drop out of the conference and be able to kick the last user that joined the conference. I believe that I can do this using:MeetMeAdmin(confno|k|userno)keeping track of the confno is easy since I created it,but I don't know how to determine the user number of the last person that joined the conference. Is there a way to store this in a variable before they join the conference? Or perhaps a way to detect the last user to join the conferences number?Any help is appreciated.Cheers,- Simon Austin ___ --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] CDR Function - Asterisk-1.2.10
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
[EMAIL PROTECTED] wrote:
AGI
===
$res = $AGI-exec(Hangup);
$foo = ${CDR(billsec)};
myVerbose($foo); #print on CLI
$foo = ${CDR(duration)};
myVerbose($foo);
$foo = ${CDR(answer)};
myVerbose($foo);
$foo = ${CDR(start)};
myVerbose($foo);
when exected with perl
Undefined subroutine main::CDR
Im passing ${EXTEN} ${CALLERIDNUM} variables through extension table. agi
accepts them fine.
${CDR(xxx)} is a function, not a variable. Are you using a library?
- --
Cheers,
Matt Riddell
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFE8cskS6d5vy0jeVcRAkj4AJ9KPowmwU/2NkPDb6tN7Md2oQ4R4ACeKwT5
RDevNr7jS/RHV5NaO7FTbTQ=
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Re: [asterisk-users] detecting a users number using the dialplan or AGI
keeping track of the confno is easy since I created it, but I don't know how to determine the user number of the last person that joined the conference. Is there a way to store this in a variable before they join the conference? Or perhaps a way to detect the last user to join the conferences number? Maybe by listing the users in the conference and parsing the output something like : meetme list 87004 you will get an output like : User #: 1 Channel: SIP/7004-1d3f (Admin) hope this help ___ --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] 7970 'LoadID incorrect' problem
First I nor sure if you can use the 7970 with SIP. LoadID looks for me the
bootimage does not match with the applicationimage. Mabe you have to erase the
flash (I'm not sure)
here is my config for Skinny Channel
venus:/srv/tftpboot # cat XMLDefault.cnf.xml
Default
callManagerGroup
members
member priority=0
callManager
ports
ethernetPhonePort2000/ethernetPhonePort
/ports
processNodeName192.168.100.25/processNodeName
/callManager
/member
/members
loadInformation7 model=IP Phone 7970P00308000100/loadInformation7
/callManagerGroup
and the boot and applicationsimage are
P00307020100.bin P00307020100.loads P00307020100.sb2 P00307020100.sbn
Hi,
Just trying to setup my 7970 with latest SIP image (SIP70.8-0-3S)
I referenced the page
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP
And used the following as my SEPmac.cnf.xml
device
devicePool
callManagerGroup
members
member priority=0
callManager
ports
ethernetPhonePort2000/ethernetPhonePort
/ports
processNodeName/processNodeName
/callManager
/member
/members
/callManagerGroup
/devicePool
versionStamp{Jan 01 2005 00:00:00}/versionStamp
loadInformationSIP70.8-0-3S/loadInformation
addOnModules
/addOnModules
userLocale
nameEnglish_United_States/name
langCodeen/langCode
/userLocale
networkLocale/networkLocale
idleTimeout0/idleTimeout
authenticationURL/authenticationURL
directoryURL/directoryURL
idleURL/idleURL
informationURL/informationURL
messagesURL/messagesURL
proxyServerURL/proxyServerURL
servicesURL/servicesURL
/device
But I get the 'LoadID incorrect' error
How do I find the correct LoadID?
I simply reset the phone everytime with **#** in settings
Thanks
Paul
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[asterisk-users] Shared NFS or Shared MySQL for redundant secondary server?
Hey List!What are your thoughts on redundancy?? Is it best to have the two Asterisk boxes share a /etc/asterisk directroy; so if one falls out of service the other takes over or is it best to have each node pull from a shared DB?? Cheers!-- --Christopher T Aloi-- ___ --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] Shared NFS or Shared MySQL for redundant secondaryserver?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Personally I've used the shared database method previously, I've even setup a mysql cluster and had each asterisk host be a query node. SKM From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Aloi Sent: Sunday, August 27, 2006 5:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Shared NFS or Shared MySQL for redundant secondaryserver? Hey List! What are your thoughts on redundancy?? Is it best to have the two Asterisk boxes share a /etc/asterisk directroy; so if one falls out of service the other takes over or is it best to have each node pull from a shared DB?? Cheers! -- -- Christopher T Aloi -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: ENCRYPTED WITH GPG iD8DBQFE8hBAlfQsv7FBhp8RArRfAKCVVuCcF+aSpLijO2rWZPa+Len05ACg1JaL z5bCCH/cWkJIAqKxsQMtC1U= =nhUW -END PGP SIGNATURE- Personally I've used the shared database method previously, I've even setup a mysql cluster and had each asterisk host be a query node. SKM From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher AloiSent: Sunday, August 27, 2006 5:31 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] Shared NFS or Shared MySQL for redundant secondaryserver? Hey List!What are your thoughts on redundancy?? Is it best to have the two Asterisk boxes share a /etc/asterisk directroy; so if one falls out of service the other takes over or is it best to have each node pull from a shared DB?? Cheers!-- --Christopher T Aloi-- PGPexch.htm.asc Description: PGP signature ___ --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] Shared NFS or Shared MySQL for redundant secondary server?
On 17:31, Sun 27 Aug 06, Christopher Aloi wrote: Hey List! What are your thoughts on redundancy?? Is it best to have the two Asterisk boxes share a /etc/asterisk directroy; so if one falls out of service the other takes over or is it best to have each node pull from a shared DB?? What we do is: Have all the data of asterisk on a NFS share. 2 machines in master/slave setup with a heartbeat between them. If master goes down, slave mounts the nfs, takes over ip and starts asterisk. Running calls will get disconnected, but the rest will continue to work. That's enough for us. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --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] SEXY WOMAN wants to know about =Callback in within voicemail broken
Is it a bug or is it me? For the longest time I have been using the feature within voicemail to call back a number by caller ID. Never had a problem with it at all. I just updated to the latest (stable) asterisk from asterisk.org Option 3 (advanced) then 2 then 1 caller number 7347292615 and now when I try to use the feature at step 2 it says: the number I have is 73472926 It 'chopps off' the last 2 digits furthermore if I press * to cancel and then have it try again I get: the number I have from an unknown caller Did I stumble upon a bug? or is there something in the changelog that I am continuing to miss that I need to asjust my configuration for. The number read back correctly in voicemail itself and shows up correctly in the CDR. Just falls apart in the callback feature when it goes to call it back. Thanks!!! Steve ___ --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] Max number of SIP devices registered to an extension
Is there a maximum number of SIP devices that can be registered to an extension?-brandon-- Brandon GalbraithEmail: [EMAIL PROTECTED] AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost ___ --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] SEXY WOMAN wants to know about =Callback in within voicemail broken
Stop trying to con lonely nerds in to answering you questions with subjects like that Steve! Anyway, check the bug tracker, I think someone posted on this list about a week ago with the exact same problem. On 8/27/06, Steve Gladden [EMAIL PROTECTED] wrote: Is it a bug or is it me? For the longest time I have been using the feature within voicemail to call back a number by caller ID. Never had a problem with it at all. I just updated to the latest (stable) asterisk from asterisk.org Option 3 (advanced) then 2 then 1 caller number 7347292615 and now when I try to use the feature at step 2 it says: the number I have is 73472926 It 'chopps off' the last 2 digits furthermore if I press * to cancel and then have it try again I get: the number I have from an unknown caller Did I stumble upon a bug? or is there something in the changelog that I am continuing to miss that I need to asjust my configuration for. The number read back correctly in voicemail itself and shows up correctly in the CDR. Just falls apart in the callback feature when it goes to call it back. Thanks!!! Steve ___ --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 ___ --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] Re: GSM gateway and FXO ATA
On Sat, Aug 26, 2006 at 02:02:51PM -0700, Martin Joseph wrote: On 2006-08-22 01:59:09 -0700, Tomislav Parčina [EMAIL PROTECTED] said: Hi list! I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over Grandstream HT488 ATA. snip Personally I found the FXO port on the HT-488 to unworkable except as a backup for power outages. I found several problems with it. 1) serious echo issues (I have a long loop). But the OP will have a very short loop. 2) If the phone is answered on the first ring the call goes off to la la land. Explaining to users (or myself) that you need to wait for the second audible ring on the handset's before answering isn't acceptable. The user here seems to be the GSM gateway. 3) The device hangs and reboots itself occasionally. Finally something relevant. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Cannot dial out through SIP provider
Shouldn't the line:
exten = _,1,Dial(SIP/[EMAIL PROTECTED],,)
be:
exten = _,1,Dial(SIP/[EMAIL PROTECTED],,)
note the .dk in the second one...
Also I don't see a register line in your sip.conf. In the [general]
section I would have expected something like:
register=a number:a password@musimi.dk/musimi
I mention this because I had a problem where the domain name didn't
resolve, so I had to change the register line to use a dotted IP
address like this:
register=a number:a password@999.999.999.999/musimi
Don't laugh, it's the only way I could get it to work!!
Yours,
H
On 8/27/06, Henrik Woffinden [EMAIL PROTECTED] wrote:
Hi,
I'm running Asterisk 1.2.10 bristuffed.
Asterisk is registring perfectly against my provider (musimi.dk), and
incoming calls comes in and are routed fine to either internal ZAP
(ISDN BRI) and/or SIP.
But
I can't dial out via SIP (musimi)
sip.conf:
[musimi]
type=friend
host=musimi.dk
username=
fromuser=
secret=xx
domain=musimi.dk
fromdomain=musimi.dk
context=from-sip
;nat=yes
;canreinvite=no
insecure=very
dtmfmode=rfc2833
[]
type=friend
context=internal
username=
secret=
host=dynamic
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=ulaw
callerid=Henrik Woffinden
nat=yes
qualify=yes
insecure=very
;[EMAIL PROTECTED]
extensions.conf:
[internal]
;exten = _,1,Dial(Zap/g1/${EXTEN},,)
exten = _,1,Dial(SIP/[EMAIL PROTECTED],,)
exten = _,n,Hangup
If I want to dial out via ISDN (Zap which is commented out above), then
it works ok, but via SIP I get the following error message (my own
number is and the number I dial is - which is a normal
mobile):
-- Registered SIP '' at 192.168.9.9 port 29796 expires 3600
-- Executing Dial(SIP/-09f2eb28, SIP/[EMAIL PROTECTED]||) in new stack
-- Called [EMAIL PROTECTED]
Aug 27 16:38:41 NOTICE[23387]: chan_sip.c:9845 handle_response_invite:
Failed to authenticate on INVITE to 'Henrik Woffinden
sip:[EMAIL PROTECTED];tag=as06ed5480'
-- SIP/musimi-09f34188 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup(SIP/-09f2eb28, ) in new stack
== Spawn extension (internal, , 2) exited non-zero on
'SIP/-09f2eb28'
I hope somebody can tell me what I'm doing wrong here.
--
Med venlig hilsen / Best regards,
Henrik Woffinden
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Re: [asterisk-users] Cannot dial out through SIP provider
Woops, sorry the first part of my response is wrong: Shouldn't the line: exten = _,1,Dial(SIP/[EMAIL PROTECTED],,) be: exten = _,1,Dial(SIP/[EMAIL PROTECTED],,) note the .dk in the second one... What I said here is incorrect, looks to me you have it right. You may still want to investigate the register command in sip.conf and using an IP address rather than a domain name... H ___ --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] Trixbox – Called party can't hangup
Hello, I apologise if this has been covered on this list in the past but I have been searching for a couple of weeks for a solution and have not yet come across one. I have set up a basic click to call service on my trixbox. The service operates by creating a call file and then once in the dialplan trixbox will call the other party. Both calls are outbound calls which are routed through my voip provider using IAX2. I do not have any phones or phone lines connected to the trixbox. The issue I'm having is with trixbox being the initiator of both calls. As far as I'm aware the Australian default is to only allow the initiator of the call to be able to terminate the call (Unless the called party is behind a pabx or on a mobile). One of the called parties has to have the phone on the hook for 90 seconds before Telstra will disconnect the call. My question is there away around this? I've done a slight work around where any party can press * to end the call but was hoping for an option that would simply let them hangup the phone. Is it possible for trixbox to detect when they have put the phone on the hook? I have read about reversing the line polarity but do not think that will help since the calls are going through a voip provider instead of directly through PSTN. Is it possible to get trixbox to detect 10 seconds of silence and hang up on a bridge call? Any ideas on the matter would be greatly appreciated. Allan. ___ --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] detecting a users number using the dialplan orAGI
You can parse the Variable BEFORE sending to the conf.
Ie:
Exten = _8700X,1,Set(${DB(conf${EXTEN}/lastin)=${CHANNEL})
It will always be the last one in.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: Sunday, August 27, 2006 4:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] detecting a users number using the
dialplan
orAGI
keeping track of the confno is easy since I created it,
but I don't know how to determine the user number of the last person
that
joined the conference.
Is there a way to store this in a variable before they join the
conference?
Or perhaps a way to detect the last user to join the conferences
number?
Maybe by listing the users in the conference and parsing the output
something like : meetme list 87004
you will get an output like :
User #: 1 Channel: SIP/7004-1d3f (Admin)
hope this help
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Re: [asterisk-users] Cannot dial out through SIP provider
- Original Message -
From: Henrik Woffinden [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, August 27, 2006 11:50 AM
Subject: [asterisk-users] Cannot dial out through SIP provider
Hi,
I'm running Asterisk 1.2.10 bristuffed.
Asterisk is registring perfectly against my provider (musimi.dk), and
incoming calls comes in and are routed fine to either internal ZAP
(ISDN BRI) and/or SIP.
But
I can't dial out via SIP (musimi)
sip.conf:
[musimi]
type=friend
host=musimi.dk
username=
fromuser=
secret=xx
domain=musimi.dk
fromdomain=musimi.dk
context=from-sip
;nat=yes
;canreinvite=no
insecure=very
dtmfmode=rfc2833
[]
type=friend
context=internal
username=
secret=
host=dynamic
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=ulaw
callerid=Henrik Woffinden
nat=yes
qualify=yes
insecure=very
;[EMAIL PROTECTED]
extensions.conf:
[internal]
;exten = _,1,Dial(Zap/g1/${EXTEN},,)
exten = _,1,Dial(SIP/[EMAIL PROTECTED],,)
exten = _,n,Hangup
If I want to dial out via ISDN (Zap which is commented out above), then
it works ok, but via SIP I get the following error message (my own
number is and the number I dial is - which is a normal
mobile):
-- Registered SIP '' at 192.168.9.9 port 29796 expires 3600
-- Executing Dial(SIP/-09f2eb28, SIP/[EMAIL PROTECTED]||) in new
stack
-- Called [EMAIL PROTECTED]
Aug 27 16:38:41 NOTICE[23387]: chan_sip.c:9845 handle_response_invite:
Failed to authenticate on INVITE to 'Henrik Woffinden
sip:[EMAIL PROTECTED];tag=as06ed5480'
-- SIP/musimi-09f34188 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup(SIP/-09f2eb28, ) in new stack
== Spawn extension (internal, , 2) exited non-zero on
'SIP/-09f2eb28'
I hope somebody can tell me what I'm doing wrong here.
Your sip provider is rejecting the call. This can be for many reasons. Bad
user/id pass, no credit left on acct., not using proper syntax etc. Look at
thier site and see how they want you to send the call to them (i.e.with the
+ sign before the number or maybe add or remove a 0)
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Re: [asterisk-users] Prompts recording for Asterisk
snip 2) What are the best sources (cost effective) to get prompts recorded. /snip I would go with allison. She is the one that did all the voice files that you currently have on asterisk. So if you use her for your prompts you will have the same voice thru out ur PBX. A client of mine just used her for his entire pbx (total of 12 clips i believe ranging in sizes). The price was $75.00 ___ --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] how to enable REACHABLE/UNREACHABLE messages in logs
Hello. I'm trying to evaluate my path to several voip providers, so I set qualify=400 in iax.conf. But, I'm not seeing any REACHABLE/UNREACHABLE or LAG messages in the logs. Is there a logging option to set so these will show up? Also, how often does asterisk do a qualify check. Thanks, Cliff -- === Cliff Brake http://bec-systems.com ___ --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] CDR Function - Asterisk-1.2.10
Hi Matt,
Thanks for the information.
Correct syntax for calling the CDR function in the AGI would be a great
help.
I have tried $foo = $AGI-exec(${CDR(xxx)});
and
$foo = $AGI-${CDR(xxx)};
none of the above works.
then i tried this:
$AGI-verbose(CDR(billsec));
$AGI-verbose(CDR(duration));
$AGI-verbose(CDR(end));
The result was = Can't locate object method CDR via package
Asterisk::AGI
And I tried this:
$AGI-verbose(${CDR(billsec)});
$AGI-verbose(${CDR(duration)});
$AGI-verbose(${CDR(end)});
got this error Undefined subroutine main::CDR called.
John
Original Message:
-
From: Matt Riddell (IT) [EMAIL PROTECTED]
Date: Sun, 27 Aug 2006 18:41:08 +0200
To: [EMAIL PROTECTED], asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CDR Function - Asterisk-1.2.10
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
[EMAIL PROTECTED] wrote:
AGI
===
$res = $AGI-exec(Hangup);
$foo = ${CDR(billsec)};
myVerbose($foo); #print on CLI
$foo = ${CDR(duration)};
myVerbose($foo);
$foo = ${CDR(answer)};
myVerbose($foo);
$foo = ${CDR(start)};
myVerbose($foo);
when exected with perl
Undefined subroutine main::CDR
Im passing ${EXTEN} ${CALLERIDNUM} variables through extension table. agi
accepts them fine.
${CDR(xxx)} is a function, not a variable. Are you using a library?
- --
Cheers,
Matt Riddell
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