Re: [asterisk-users] CDR Issue
On Tue, 04 Sep 2012 14:24:54 +0530 Parveen Lamba pla...@tekege.com wrote: When I call to 0X number using sip/test, CDR is created between sip and dahdi channel. [...] Here disposition is always answered whether I attend or reject the call. This is normal with analog lines. When you place a call on the line, it's answered immediately because there's no signalling. Call progress information is given solely by tones in the audio. If you want signalling, you have to use digital lines (like ISDN or SIP). -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR Issue
Hi,
I have configured asterisk with Sangoma analog card. Outbound and
Inbound calls are working fine. I have issue with CDR for outbound call.
When I call to 0X number using sip/test, CDR is created between
sip and dahdi channel.
Here is CDR and CEL :-
+-++-+-+-+-++-+-+--+-+-+--+-+---+---+
| calldate| clid | src | dst |
dcontext| channel | dstchannel | lastapp |
lastdata| duration | billsec | disposition | amaflags |
accountcode | uniqueid | userfield |
+-++-+-+-+-++-+-+--+-+-+--+-+---+---+
| 2012-09-05 02:11:25 | sip/test 101 | 101 | 0XX |
test | SIP/test-0024 | DAHDI/3-1 | Dial|
dahdi/g0/0XX,20 | 48 | 33 | ANSWERED|
3 | | 1346825485.69 | |
+-++-+-+-+-++-+-+--+-+-+--+-+---+---+
Here disposition is always answered whether I attend or reject the call.
eventtype eventtime CALLERID(name) CALLERID(num) CALLERID(ANI)
CALLERID(RDNIS) CALLERID(DNID) CHANNEL(exten) CHANNEL(context)
CHANNEL(channame) CHANNEL(appname) CHANNEL(appdata)
CHANNEL(amaflags) CHANNEL(accountcode) CHANNEL(uniqueid)
CHANNEL(linkedid) BRIDGEPEER CHANNEL(userfield) userdeftype eventextra
CHAN_START 2012-09-05 02:11:25
101
s testSIP/test-0024
3
1346825485.69 1346825485.69
ANSWER 2012-09-05 02:11:37 sip/test101 101
testSIP/test-0024
3
1346825485.69 1346825485.69
CHAN_START 2012-09-05 02:11:37
s from-zaptel DAHDI/3-1
3
1346825497.71346825485.69
ANSWER 2012-09-05 02:11:40 sip/testXX
9717330017 from-zaptel DAHDI/3-1 AppDial
(Outgoing Line) 3
1346825497.71346825485.69
BRIDGE_START2012-09-05 02:11:40 sip/test101 101
s macro-std SIP/test-0024 Dial
dahdi/g0/0XX,20 3
1346825485.69 1346825485.69 DAHDI/3-1
BRIDGE_END 2012-09-05 02:12:13 sip/test101 101
s macro-std SIP/test-0024 Dial
dahdi/g0/0XX,20 3
1346825485.69 1346825485.69 DAHDI/3-1
HANGUP 2012-09-05 02:12:13 sip/testXX
macro-std DAHDI/3-1 AppDial (Outgoing Line)
3
1346825497.71346825485.69
16,SIP/test-0024,
CHAN_END2012-09-05 02:12:13 sip/testXX
macro-std DAHDI/3-1 AppDial (Outgoing Line)
3
1346825497.71346825485.69
HANGUP 2012-09-05 02:12:13 sip/test101 101
9717330017 testSIP/test-0024
3
1346825485.69 1346825485.69
16,SIP/test-0024,ANSWER
CHAN_END2012-09-05 02:12:13 sip/test101 101
9717330017 testSIP/test-0024
3
1346825485.69 1346825485.69
LINKEDID_END2012-09-05 02:12:13 sip/test101 101
9717330017 testSIP/test-0024
3
1346825485.69 1346825485.69
and here is my dial plan:
[test]
exten = _XXX,1,Dial(dahdi/g0/${EXTEN:})
Could anyone please tell me where I am doing mistake.
Any help will be appreciable.
Thanks
Parveen
--
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[asterisk-users] CDR issue - Problem logging CDR(userfield) in Master.csv
Dear all,
I am having an issue with CDR logging. What I want to do is log jitter
variable from RTPAUDIOQOS module into Master.csv at the end of each call.
I am using asterisk version 1.4.26. For CDR purposes, I am using
cdr_custom, and the content of my cdr_custom.conf is the following:
[mappings]
Master.csv =
${CDR(dstchannel)},${CDR(clid)},${CDR(cid-num)},${CDR(dst)},${CDR(start)},${CDR(billsec)},${CDR(disposition)},${CDR(userfield)}
In my dialplan, right before my Hangup() call, I have put the following
(am using AEL, but I guess this is irrelevant)
Set(JITTER=${CUT(RTPAUDIOQOS,\;,4)});
Set(CDR(userfield)=${CUT(JITTER,\=,2)});
I make an outgoing call with asterisk debugging mode on. When I hangup
the call, I see the following in the asterisk cli:
Executing [sw-41-ANSWER@macro-handle_dialstatus:10]
Set(SIP/FXS1-001666f8, JITTER=rxjitter=0.004785) in new stack
- Executing [sw-41-ANSWER@macro-handle_dialstatus:11]
Set(SIP/FXS1-001666f8, CDR(userfield)=0.004785) in new stack
which means that my commands should work.
However, when I check the contents of my Master.csv file, the userfield
variable is EMPTY :-(
SIP/line1-00141f30,FXS1 FXS1,,210999,2011-01-28
13:21:16,39,ANSWERED,
What could be the problem?
Best Regards and thank you in advance,
Athanasia
--
--
Athanasia Tsertou
Software Engineer
Gennet s.a.
2, Mesogeion Ave., Athens Tower
Athens GR-115 27
Tel: +30 210 74 58 435
Fax: +30 210 74 58 481
Mob: +30 697 87 85 883
e-mail: atser...@gennetsa.com
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Re: [asterisk-users] CDR issue - Problem logging CDR(userfield) in Master.csv
On Friday 28 January 2011 05:34:21 Athanasia Tsertou wrote:
In my dialplan, right before my Hangup() call, I have put the following
(am using AEL, but I guess this is irrelevant)
Set(JITTER=${CUT(RTPAUDIOQOS,\;,4)});
Set(CDR(userfield)=${CUT(JITTER,\=,2)});
Did you put it in the h extension? That is where it needs to be executed
in order for it to go into the CDR. If it's not in the h extension,
you're updating the CDR after it has already been posted.
--
Tilghman
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[asterisk-users] CDR issue
Hello! Ive an issue whit CDR using asterisk 1.4.23.1. Ive configured mysql to store cdr information, but, while I put into cdr_mysql.conf the field userfield=1 and doing a query I found that this field is empty in the cdr table. On the other hand I cant find records in the cdr table that show me calls generated through AMI using Originate Action, thats calls are not stored in the CDR, but I dont know why if I calls from pstn are stored whitout problems. Parameters that I send are: Channe:SIP/$CHANNEL CallerID:listener Application:ChanSpy' Data:SIP/AgentPhone. Variable:CDR(userfield)=listened Async=yes Thanks for any idea about that Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. mailto:ggonza...@despegar.com ggonza...@despegar.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR issue
Gustavo A Gonzalez escribió: Hello! I've an issue whit CDR using asterisk 1.4.23.1. I've configured mysql to store cdr information, but, while I put into cdr_mysql.conf the field 'userfield=1' and doing a query I found that this field is empty in the cdr table. On the other hand I can't find records in the cdr table that show me calls generated through AMI using Originate Action, that's calls are not stored in the CDR, but I don't know why if I calls from pstn are stored whitout problems. Parameters that I send are: Channe:SIP/$CHANNEL CallerID:listener Application:ChanSpy' Data:SIP/AgentPhone. Variable:CDR(userfield)=listened Async=yes Well, there's two things to look around here. First, the variable en the AMI Originate action lets you set channel variables on to the channel of the originated call, not onto the spied (and already established) call. Second, if you can't see the originated calls on the CDR, try setting the option unanswered=yes on cdr.conf and issue a cdr reload command on the CLI. You can check with cdr status that this option is set, that tells asterisk to maintain a CDR for *every* call attempt that goes through it. Then you can check if the originated calls show up on your CDR database. Cheers, Thanks for any idea about that *Gustavo A. González* Dto. de Infraestructura Despegar.com, Inc. ggonza...@despegar.com mailto:ggonza...@despegar.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57 1)6500816 Móvil: (+57)3138873587 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR issue
Thanks!Ive solve the issue setting: unanswered=yes on cdr.conf . Cheers! Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. ggonza...@despegar.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR issue need help
gud day, hope this is not OT. currently encountering problem with ATA FXO FAS problem especially on CDR. the scenario is dis: ATA of grandstream and sipura, send 183 and 200 simultaneously even if the FXO port is not yet ringing. since 200 was receive already then CDR starts the timer. not good for a prepaid system research: all documents that i have read points me to a blank wall. possible workaround: upon receiving of 200 delay for a while so that CDR does not start immediately. prepaid customer is not that much. help: im asking the developers if this is possible? how can i do this.. all help is welcome, regards arvin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr issue
Dear all: hi, i face some problem in the mysql cdr module. Here is my situation, hope you all give some comments. (cdr) User --- Asterisk -- gateway (IVR) --- (B party) (call access number) (enter destination ) For the case above, i try to use asterisk to bypass the gateway process which mean asterisk dial the access and send dtmf to the gateway so that the user no need to call the access number , they just need to enter the destination number. It work fine, but i face problem for the cdr. The cdr is consider "CONNECTED" when the gateway answer the call from asterisk. But what i need is "CONNECTED" only if the B party is pick up the phone. Now the B party didnt pick up the phone, the call still consider "CONNECTED" due to the gateway already answer the call from asterisk, and even the B party pick up the phone, the duration still not accurate coz the counter start count while the gateway answer the call but not counted while the B party answer the call. So is there any way to solve this cdr issue or what i can do ? Please kindly advise. I got one idea which is when the B party answer call, the gateway will send the dtmf tone back to asterisk, then theasterisk can fire theCONNECTED event, but idont know how to do it in asterisk, please help.. Thanks in advance Regards HSL ___ 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
