[asterisk-users] Billsec 0 when using call file to Local channel via cdr_adapative_odbc
Hi, all Sorry that forgot add mail subject last one. I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database "mydatabase") via cdr_adaptive_odbc. The "SIP/A221" is another asterisk machine named it "Elastix24". I have two BIG QUESTIONs about cdr_adaptive_odbc. First, I have answered call from Elastix24 and I can listen the music file played from Asterisk11. In another word, this call should be answered and its billsec is greater than 0. Second, if I don't want to use forkcdr(), how to config it and I can get another cdr record that call from SIP/A221(Elastix24) to my Exten:77? I know that the outgoing file will make a call to Local Channel and try to Dial SIP/A221. If it answered, this old channel should be hangup and generate another new channel to connect to Extension:77(my callback exten). I can't find two cdr records in mycdr table. mysql> select * from gvl_cdr; +-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++ | calldate| clid | src | dst | dcontext| channel | dstchannel| lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | userfield | uniqueid | linkedid | sequence | peeraccount | phoneno | callerid | userid | +-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++ | 2014-01-08 14:37:01 | | |77 | from-internal-out-7 | Local/77@from-internal-out-7-;2 | SIP/A221- | Dial| SIP/A221/77,30 | 17 | 0 | ANSWERED| 3 | | | 1389163021.1 | 1389163021.0 | 1| | 77 | | 7 | Even I try to add ForkCDR or ResetCDR. The billsec is 0 in other record(the 3th one). mysql> select * from gvl_cdr; +-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++ | calldate| clid | src | dst | dcontext| channel| dstchannel| lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | userfield | uniqueid | linkedid | sequence | peeraccount | phoneno | callerid | userid | +-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++ | 2014-01-08 14:34:04 || | 77| from-internal-out-7 | Local/77@from-internal-out-7-;2| SIP/A221- | Dial| SIP/A221/77,30| 15 | 0 | ANSWERED|3 | | | 1389162844.1 | 1389162844.0 | 1| | 77 | | 7 | | 2014-01-08 14:34:04 | "device" <1000>| 1000| 77| from-6 | Local/77@from-internal-out-7-;1| | ForkCDR | | 20 | 5 | ANSWERED|3 | | | 1389162844.0 | 1389162844.0 | 0| | 77 | | 7 | | 2014-01-08 14:34:24 | "device" <77> | 77 | 77| from-6 | Local/77@from-internal-out-7-;1| | Read| CALLBACK,custom-gvl/2,1,s,1,3 |0 | 0 | NO ANSWER |3 | | | 1389162844.0 | 1389162844.0 | 3| | | | 0 | - /var/spool/asterisk/outgoing/77.call Channel:Local/77@from-internal-out-7 WaitTime:30 Context:from-6 Extension:77 Priority:1 Set:CLID= Set:EXT=77 Set:USERID=7 ---
[asterisk-users] (no subject)
Hi, all I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database "mydatabase") via cdr_adaptive_odbc. The "SIP/A221" is another asterisk machine named it "Elastix24". I have two BIG QUESTIONs about cdr_adaptive_odbc. First, I have answered call from Elastix24 and I can listen the music file played from Asterisk11. In another word, this call should be answered and its billsec is greater than 0. Second, if I don't want to use forkcdr(), how to config it and I can get another cdr record that call from SIP/A221(Elastix24) to my Exten:77? I know that the outgoing file will make a call to Local Channel and try to Dial SIP/A221. If it answered, this old channel should be hangup and generate another new channel to connect to Extension:77(my callback exten). I can't find two cdr records in mycdr table. mysql> select * from gvl_cdr; +-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++ | calldate| clid | src | dst | dcontext| channel | dstchannel| lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | userfield | uniqueid | linkedid | sequence | peeraccount | phoneno | callerid | userid | +-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++ | 2014-01-08 14:37:01 | | |77 | from-internal-out-7 | Local/77@from-internal-out-7-;2 | SIP/A221- | Dial| SIP/A221/77,30 | 17 | 0 | ANSWERED| 3 | | | 1389163021.1 | 1389163021.0 | 1| | 77 | | 7 | Even I try to add ForkCDR or ResetCDR. The billsec is 0 in other record(the 3th one). mysql> select * from gvl_cdr; +-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++ | calldate| clid | src | dst | dcontext| channel| dstchannel| lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | userfield | uniqueid | linkedid | sequence | peeraccount | phoneno | callerid | userid | +-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++ | 2014-01-08 14:34:04 || | 77| from-internal-out-7 | Local/77@from-internal-out-7-;2| SIP/A221- | Dial| SIP/A221/77,30| 15 | 0 | ANSWERED|3 | | | 1389162844.1 | 1389162844.0 | 1| | 77 | | 7 | | 2014-01-08 14:34:04 | "device" <1000>| 1000| 77| from-6 | Local/77@from-internal-out-7-;1| | ForkCDR | | 20 | 5 | ANSWERED|3 | | | 1389162844.0 | 1389162844.0 | 0| | 77 | | 7 | | 2014-01-08 14:34:24 | "device" <77> | 77 | 77| from-6 | Local/77@from-internal-out-7-;1| | Read| CALLBACK,custom-gvl/2,1,s,1,3 |0 | 0 | NO ANSWER |3 | | | 1389162844.0 | 1389162844.0 | 3| | | | 0 | - /var/spool/asterisk/outgoing/77.call Channel:Local/77@from-internal-out-7 WaitTime:30 Context:from-6 Extension:77 Priority:1 Set:CLID= Set:EXT=77 Set:USERID=7 -- /etc/asterisk/extensions.conf lists be
Re: [asterisk-users] Asterisk 1.6.2.x Keeping NAT Alive
On Tue, Jan 7, 2014 at 4:32 PM, wrote: > I'm running Asterisk 1.6.2.10, and I'm having the issue described here; > https://issues.asterisk.org/jira/browse/ASTERISK-15721 > > The last note says Patch "heap-fix.rev2.diff" was uploaded - was this > Patch released, I don't see a reference for it in the thread? Anyone > download patch and install it? > The issue you reference shows that the patch was committed to the v1.6.2 branch with SVN revision -r261498. The commit message mentions the patch file but it does not appear to be attached to the issue. You can use SVN to generate a diff file of the committed change: svn diff -c261498 http://svn.asterisk.org/svn/asterisk > change.patch Richard -- _ -- 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] Asterisk NAT friendly settings
I'm asking about this scenario: Asterisk(public IP) <--> Internet <--> Router (public IP) <--> SIP client (private IP and NAT) What settings in sip.conf will give this the best fighting chance of working? We already have nat=force_rport,comedia -- _ -- 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] Asterisk 1.6.2.x Keeping NAT Alive
I'm running Asterisk 1.6.2.10, and I'm having the issue described here;https://issues.asterisk.org/jira/browse/ASTERISK-15721The last note says Patch "heap-fix.rev2.diff" was uploaded - was this Patch released, I don't see a reference for it in the thread? Anyone download patch and install it?Kind Regards,Fred Robinson -- _ -- 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
Re: [asterisk-users] Lot of voice cut
Sometimes you need to disable B-channel restarts, see chan_dahdi.conf.sample -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitesh Thakkar Sent: Tuesday, January 07, 2014 2:35 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Lot of voice cut > What's the config look like on the other side? Is the D-Chan on some random > channel? Or 24 and 48 as configured? It is 24 and 48. Kindly find the system.conf as below: # Span 1: WCTE2/0/1 "WCTE23X (PCI) Card 0 Span 1" (MASTER) ESF/B8ZS ClockSource span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2: WCTE2/0/2 "WCTE23X (PCI) Card 0 Span 2" ESF/B8ZS span=2,2,0,esf,b8zs # termtype: te bchan=25-47 dchan=48 echocanceller=mg2,25-47 # Global data loadzone = us defaultzone = us == What could be the solution? Regards, Mitesh Thakkar -- _ -- 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
Re: [asterisk-users] Lot of voice cut
> What's the config look like on the other side? Is the D-Chan on some random > channel? Or 24 and 48 as configured? It is 24 and 48. Kindly find the system.conf as below: # Span 1: WCTE2/0/1 "WCTE23X (PCI) Card 0 Span 1" (MASTER) ESF/B8ZS ClockSource span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2: WCTE2/0/2 "WCTE23X (PCI) Card 0 Span 2" ESF/B8ZS span=2,2,0,esf,b8zs # termtype: te bchan=25-47 dchan=48 echocanceller=mg2,25-47 # Global data loadzone= us defaultzone = us == What could be the solution? Regards, Mitesh Thakkar-- _ -- 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
Re: [asterisk-users] Lot of voice cut
Well, Not to point out the obvious. But it's complaining about the D-Channel not being in service. What's the config look like on the other side? Is the D-Chan on some random channel? Or 24 and 48 as configured? For reference. This is what one of my PRI's looks like. ]# service dahdi status ### Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) B8ZS/ESF 1 PRIClear (In use) (SWEC: MG2) (EC: VPM450M) 2 PRIClear (In use) (SWEC: MG2) 3 PRIClear (In use) (SWEC: MG2) 4 PRIClear (In use) (SWEC: MG2) 5 PRIClear (In use) (SWEC: MG2) 6 PRIClear (In use) (SWEC: MG2) 7 PRIClear (In use) (SWEC: MG2) 8 PRIClear (In use) (SWEC: MG2) 9 PRIClear (In use) (SWEC: MG2) 10 PRIClear (In use) (SWEC: MG2) 11 PRIClear (In use) (SWEC: MG2) 12 PRIClear (In use) (SWEC: MG2) 13 PRIClear (In use) (SWEC: MG2) 14 PRIClear (In use) (SWEC: MG2) 15 PRIClear (In use) (SWEC: MG2) 16 PRIClear (In use) (SWEC: MG2) 17 PRIClear (In use) (SWEC: MG2) 18 PRIClear (In use) (SWEC: MG2) 19 PRIClear (In use) (SWEC: MG2) 20 PRIClear (In use) (SWEC: MG2) 21 PRIClear (In use) (SWEC: MG2) 22 PRIClear (In use) (SWEC: MG2) 23 PRIClear (In use) (SWEC: MG2) 24 PRIHDLCFCS (In use) Nick Olsen Network Operations (855) FLSPEED x106 From: "Mitesh Thakkar" Sent: Tuesday, January 07, 2014 2:55 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Lot of voice cut I have added some more information - service dahdi status and system.conf including error at http://pastebin.com/a0Zg881P Kindly have a look. Regards. On Wednesday 08 Jan 2014 12:59:39 AM Mitesh Thakkar wrote: > Hello, > > I am facing lot of voice cut after installing wctw235b card. > > I am using libpri-1.4.14, dahdi-linux-complete-2.8.0.1+2.8.0, > asterisk-11.7.0. > > I see error (http://pastebin.com/yJEgnnq7) on console. > > Can anyone help me in this please. > > Regards. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Lot of voice cut
I have added some more information - service dahdi status and system.conf including error at http://pastebin.com/a0Zg881P Kindly have a look. Regards. On Wednesday 08 Jan 2014 12:59:39 AM Mitesh Thakkar wrote: > Hello, > > I am facing lot of voice cut after installing wctw235b card. > > I am using libpri-1.4.14, dahdi-linux-complete-2.8.0.1+2.8.0, > asterisk-11.7.0. > > I see error (http://pastebin.com/yJEgnnq7) on console. > > Can anyone help me in this please. > > Regards. -- _ -- 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] Lot of voice cut
Hello, I am facing lot of voice cut after installing wctw235b card. I am using libpri-1.4.14, dahdi-linux-complete-2.8.0.1+2.8.0, asterisk-11.7.0. I see error (http://pastebin.com/yJEgnnq7) on console. Can anyone help me in this please. Regards. -- _ -- 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] Asterisk CallCompletion issue (NO_CORE_INSTANCE)
Hello All,
I have some problem on asterisk call completion feature, Every time I want to
execute the application(By pressing 6) , I got the following error:
"NO_CORE_INSTANCE"
So, I was in shocked sometimes without making any changes on the dialplan coed
problem solved.
But in most of the time the problem kept persisting.
Now I need some helps to solved the problem permananely?
Thanks for your help.
exten => 6,1,CallCompletionRequest()
exten => 6,n,playback(beep)
exten => 6,n,verbose(${CC_REQUEST_RESULT})
exten => 6,n,goto(${CC_REQUEST_RESULT},1)
exten => SUCCESS,1,verbose(${CC_REQUEST_RESULT})
exten => FAIL,1,verbose(${CC_REQUEST_REASON})same => n,hangup()
exten => 7,1,CallCompletionCancel()
exten => 7,n,playback(beep)
exten => 7,n,hangup()
Regard
Sami--
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