[asterisk-users] [Asterisk 1.6] Mysql cdr addon doen't write full channel infomation when disposition is Failed
Hi All, Currently i'm facing with a cdr issue, When i originate a call (outbound call) to uncorrect/unregistration user, asterisk inform me that call was failed but in mysl-cdr (cdr-csv also) records. Here are 2 samples +-+--+-+-+--+-++--+-+---+ | calldate| clid | src | dst | dcontext | channel | dstchannel | duration | disposition | userfield | +-+--+-+-+--+-++--+-+---+ | 2013-03-30 11:01:20 | | | s | default | | |0 | FAILED | | | 2013-03-22 08:45:00 | | | 777 | from-avc | SIP/083777-0013 | | 19 | ANSWERED| 16| Thank and Appreciate if any social experiences can help me on this. BRs. -- TrungND -- _ -- 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] extensions.conf / test DID
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI sip show registry
Hostdnsmgr Username Refresh
StateReg.Time
sip3.voipvoip.com:5060 N 1112530146 105
Registered Mon, 08 Apr 2013 06:02:09
1 SIP registrations.
Asterisk*CLI
Here is the dial plan:
[incoming]
exten = 17036361355,1,Playback(beep)
exten = 17036361355,2,SayDigits(${EXTEN})
exten = 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle mobile phone.
;exten = s,1,Answer()
;exten = s,n,Dial(SIP/17037171234,150,r,t,)
[general]
register =1112530146:albany!@#1...@sip3.voipvoip.com/1112530146
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.1.10
srvlookup=no
;context=incoming
; The SIP provider
[voipvoip.com]
canreinvite=no
username=1112530146
fromuser=1112530146
secret=albany!@#123
context=incoming
type=friend
fromdomain=s...@voipvoip.com
host=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=force_rport
insecure=port,invite
Thoughts please?I think something to do w/ incoming is incorrect.
--
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Re: [asterisk-users] extensions.conf / test DID
On Monday 08 April 2013, Thomas Perron wrote:
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI sip show registry
Hostdnsmgr Username Refresh
StateReg.Time
sip3.voipvoip.com:5060 N 1112530146 105
Registered Mon, 08 Apr 2013 06:02:09
1 SIP registrations.
Asterisk*CLI
Here is the dial plan:
[incoming]
exten = 17036361355,1,Playback(beep)
exten = 17036361355,2,SayDigits(${EXTEN})
exten = 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle mobile phone.
;exten = s,1,Answer()
;exten = s,n,Dial(SIP/17037171234,150,r,t,)
[general]
register =1112530146:albany!@#1...@sip3.voipvoip.com/1112530146
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.1.10
srvlookup=no
;context=incoming
; The SIP provider
[voipvoip.com]
canreinvite=no
username=1112530146
fromuser=1112530146
secret=albany!@#123
context=incoming
type=friend
fromdomain=s...@voipvoip.com
host=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=force_rport
insecure=port,invite
Thoughts please?I think something to do w/ incoming is incorrect.
You only have one extension, 17036361355 in the [incoming] context in your
dialplan. Are you sure that 17036361355 is exactly what the SIP provider
are actually sending to your end ?
I'd put an s extension with a NoOp(${EXTEN}) in there, just to catch the
actual extension number they were sending.
--
AJS
Answers come *after* questions.
--
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asterisk-users mailing list
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Re: [asterisk-users] extensions.conf / test DID
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI sip show registry
Hostdnsmgr Username Refresh
StateReg.Time
sip3.voipvoip.com:5060 N 1112530146 105
Registered Mon, 08 Apr 2013 06:02:09
1 SIP registrations.
Asterisk*CLI
Here is the dial plan:
[incoming]
exten = 17036361355,1,Playback(beep)
exten = 17036361355,2,SayDigits(${EXTEN})
exten = 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle mobile phone.
;exten = s,1,Answer()
;exten = s,n,Dial(SIP/17037171234,150,r,t,)
[general]
register =1112530146:albany!@#1...@sip3.voipvoip.com/1112530146
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.1.10
srvlookup=no
;context=incoming
; The SIP provider
[voipvoip.com]
canreinvite=no
username=1112530146
fromuser=1112530146
secret=albany!@#123
context=incoming
type=friend
fromdomain=s...@voipvoip.com
host=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=force_rport
insecure=port,invite
Thoughts please?I think something to do w/ incoming is incorrect.
[incoming]
exten = 17036361355,1,Playback(beep)
exten = 17036361355,2,SayDigits(${EXTEN})
exten = 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle mobile phone.
;exten = s,1,Answer()
;exten = s,n,Dial(SIP/17037171234,150,r,t,)
As well doesn't the Goto need to closing )?
--
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Re: [asterisk-users] extensions.conf / test DID
On Mon, Apr 8, 2013 at 4:26 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote:
On Monday 08 April 2013, Thomas Perron wrote:
I am trying to make sure my DID and SIP account details are working
properly and engaging the extensions.conf and dial plan.
I have a successful SIP session registered:
Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
Asterisk*CLI sip show registry
Hostdnsmgr Username Refresh
StateReg.Time
sip3.voipvoip.com:5060 N 1112530146 105
Registered Mon, 08 Apr 2013 06:02:09
1 SIP registrations.
Asterisk*CLI
Here is the dial plan:
[incoming]
exten = 17036361355,1,Playback(beep)
exten = 17036361355,2,SayDigits(${EXTEN})
exten = 17036361355,3,Goto(testdtmf|s|1
;Ring on Elle mobile phone.
;exten = s,1,Answer()
;exten = s,n,Dial(SIP/17037171234,150,r,t,)
[general]
register =1112530146:albany!@#1...@sip3.voipvoip.com/1112530146
registertimeout=20
context=incoming
allowoverlap=no
bindport=5060
bindaddr=192.168.1.10
srvlookup=no
;context=incoming
; The SIP provider
[voipvoip.com]
canreinvite=no
username=1112530146
fromuser=1112530146
secret=albany!@#123
context=incoming
type=friend
fromdomain=s...@voipvoip.com
host=69.90.209.57
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
nat=force_rport
insecure=port,invite
Thoughts please?I think something to do w/ incoming is incorrect.
You only have one extension, 17036361355 in the [incoming] context in
your
dialplan. Are you sure that 17036361355 is exactly what the SIP provider
are actually sending to your end ?
I'd put an s extension with a NoOp(${EXTEN}) in there, just to catch the
actual extension number they were sending.
--
AJS
Answers come *after* questions.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
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I don't think s extension will work on SIP channel. s extension is a
catch-all extension for Analog calls and Macros (reference:
https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions)
Just for the sake of testing I would have something like,
[incoming]
exten = _X.,1,NoOp(EXTENSION=${EXTEN})
exten = _X.,2,Playback(beep)
exten = _X.,3,SayDigits(${EXTEN})
exten = _X.,3,Goto(testdtmf|s|1)
;Ring on Elle mobile phone.
;exten = s,1,Answer()
;exten = s,n,Dial(SIP/17037171234,150,r,t,)
--
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Re: [asterisk-users] extensions.conf / test DID
I don't think s extension will work on SIP channel. s extension is a catch-all extension for Analog calls Console output would be useful. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] extensions.conf / test DID
On Mon, 8 Apr 2013, Thomas Perron wrote: I am trying to make sure my DID and SIP account details are working properly and engaging the extensions.conf and dial plan. If you jack up logging, you may see a message on the console like: looking for x in y where x is the extension and y is the context. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] OT - How to simulate public IPs for lab testing
Hello, Many times, I need to test in a lab Asterisk servers before sending them to customer locations. I'm currently having trouble to test SIP trunks without touching SIP configuration. So, how should I change my testing lab so that I could now test SIP trunks without modifying Asterisk server under test ? A typical set up is: Asterisk server1 under test ---SIP Router - SIP Lab's Asterisk server2 All machines (server1, router and server2) have Internet access. Router and server2 have a private address. Ideally, router should get customer's public adress (eg 1.2.3.4), server2 should also get my ITSP public address (eg 4.3.2.1) and both machines should route trafic to each other without leaving my LAN and using Internet access. What would you suggest ? 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
Re: [asterisk-users] OT - How to simulate public IPs for lab testing
2013-04-08 16:36, Olivier skrev: Hello, Many times, I need to test in a lab Asterisk servers before sending them to customer locations. I'm currently having trouble to test SIP trunks without touching SIP configuration. So, how should I change my testing lab so that I could now test SIP trunks without modifying Asterisk server under test ? A typical set up is: Asterisk server1 under test ---SIP Router - SIP Lab's Asterisk server2 All machines (server1, router and server2) have Internet access. Router and server2 have a private address. Ideally, router should get customer's public adress (eg 1.2.3.4), server2 should also get my ITSP public address (eg 4.3.2.1) and both machines should route trafic to each other without leaving my LAN and using Internet access. What would you suggest ? I often configure a router to do NAT in these cases. You can do NAT even with a public net on the inside. Configure the temporary router with the IP of the customers router for the inside, and make it a dhcp client (or whatever you use) in your LAN for the outside interface. You can make a route in your router to your ITSP-gw like this : route add -host 4.3.2.1 dev eth0 This means all traffic to 4.3.2.1 will go to dev eth0 on your router. (The device in your network with the ip 4.3.2.1 also needs to have a route back to your router for replies.) Good luck! -- Johan Wilfer -- _ -- 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] OT - How to simulate public IPs for lab testing
2013/4/8 Johan Wilfer li...@jttech.se 2013-04-08 16:36, Olivier skrev: Hello, Many times, I need to test in a lab Asterisk servers before sending them to customer locations. I'm currently having trouble to test SIP trunks without touching SIP configuration. So, how should I change my testing lab so that I could now test SIP trunks without modifying Asterisk server under test ? A typical set up is: Asterisk server1 under test ---SIP Router - SIP Lab's Asterisk server2 All machines (server1, router and server2) have Internet access. Router and server2 have a private address. Ideally, router should get customer's public adress (eg 1.2.3.4), server2 should also get my ITSP public address (eg 4.3.2.1) and both machines should route trafic to each other without leaving my LAN and using Internet access. What would you suggest ? I often configure a router to do NAT in these cases. You can do NAT even with a public net on the inside. Configure the temporary router with the IP of the customers router for the inside, and make it a dhcp client (or whatever you use) in your LAN for the outside interface. You can make a route in your router to your ITSP-gw like this : route add -host 4.3.2.1 dev eth0 This means all traffic to 4.3.2.1 will go to dev eth0 on your router. (The device in your network with the ip 4.3.2.1 also needs to have a route back to your router for replies.) Please, excuse me but I'm not sure I got your suggestion and I'm realizing I didn't correctly describe my lab set up. At the moment, the router between both servers provides Internet access to server1. That means it has one WAN interface eth0 which is on server2 side and one eth1 LAN interface which is on server1 side. Currently, this router do NAT translation for server1. Having clarified my setup, I guess your advice is to : 1. add address 1.2.3.4 to router's eth0 2. add address 4.3.2.1 to server2 interface 3. configure router to route trafic to 4.3.2.1 using server2 private address (such as ip route add 4.3.2.1/24 via 192.168.1.25) 4. configure server to route trafic to 1.2.3.4 using router private address Is this correct ? Good luck! -- Johan Wilfer -- __**__**_ -- 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-**usershttp://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
[asterisk-users] dahdi strange state error
System details: Digium Wildcard TDM410P with three extensions and one POTS line. Dual core Pentium 4 (32-bit) processor Fedora 18 Asterisk 11.2.1 DAHDI Version: 2.6.2 Echo Canceller: HWEC I recently upgraded from Asterisk 1.4, and made as few changes to the configuration files as possible. I regenerated the dahdi-channels.conf file, then I believe I had to edit this to set the contexts correctly to match the rest of my dialplan. The dialplan does seem to be working as before. The problem is that, every few days, my POTS line gets stuck. The other ports are working for intercom, but attempts to place outbound calls through the POTS line, whether from a SIP phone or one of the DAHDI extensions, get a fast busy, and this appears in the Asterisk log: [Apr 5 21:29:35] WARNING[8191][C-003e] sig_analog.c: Ring/Off-hook in strange state 6 on channel 4 I have to restart asterisk and/or dahdi to get it working again. Can someone tell me what this message means and suggest how I might prevent this from happening? Is my hardware flaky, or is there something I need to update in my configuration? Thank you, --Greg -- _ -- 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
