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Ignored DialEvent for unknown destination channel null with

January 20, 2014, 5:15 pm
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The problem appears to be in your Java class, not in Asterisk.

Statistics : Posted by logankim • on Mon Jan 20, 2014 7:15 pm • Replies 1 • Views 38
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Ignored DialEvent for unknown destination channel null with

January 20, 2014, 5:15 pm
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Hi.

After I installed asterisk 11, I got this error everytime. But everything works fine.

Ignored DialEvent for unknown destination channel null with unique id null

I thought this is because DialEvent has null in destination field when subevent is "End".

[2014-01-17 21:06:41,712 DEBUG org.asteriskjava.manager.internal.ManagerConnectionImpl] - Dispatching event:
org.asteriskjava.manager.event.DialEvent[dateReceived='Fri Jan 17 21:06:41 PST 2014',privilege='call,all',subevent='End',callerid=null,dialstatus='CANCEL',sequencenumber=null,destuniqueid=null,
srcuniqueid='1390021576.106',destination=null,dialstring=null,timestamp=null,calleridname=null,
uniqueid='1390021576.106',server=null,src='Local/s@auth-00000023;2',calleridnum=null,channel='Local/s@auth-00000023;2',systemHashcode=27029873]

[2014-01-17 21:06:41,713 ERROR org.asteriskjava.live.internal.ChannelManager]
- Ignored DialEvent for unknown destination channel null with unique id null

I think this is not a big issue.
But I feel uneasy about why org.asteriskjava.live.internal.ChannelManager log ERROR.

Statistics : Posted by logankim • on Mon Jan 20, 2014 7:15 pm • Replies 1 • Views 38
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Ignored DialEvent for unknown destination channel null with

January 20, 2014, 5:14 pm
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The problem is in you Java AMI class library, There is insufficient evidence to indicate that there is anything wrong with Asterisk. Please take this up with the developers of the class library.

Statistics : Posted by logankim • on Mon Jan 20, 2014 7:14 pm • Replies 1 • Views 38
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Ignored DialEvent for unknown destination channel null with

January 20, 2014, 5:14 pm
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Hi.

After I installed asterisk 11, I got this error everytime. But everything works fine.

Ignored DialEvent for unknown destination channel null with unique id null

I thought this is because DialEvent has null in destination field when subevent is "End".

[2014-01-17 21:06:41,712 DEBUG org.asteriskjava.manager.internal.ManagerConnectionImpl] - Dispatching event:
org.asteriskjava.manager.event.DialEvent[dateReceived='Fri Jan 17 21:06:41 PST 2014',privilege='call,all',subevent='End',callerid=null,dialstatus='CANCEL',sequencenumber=null,destuniqueid=null,
srcuniqueid='1390021576.106',destination=null,dialstring=null,timestamp=null,calleridname=null,
uniqueid='1390021576.106',server=null,src='Local/s@auth-00000023;2',calleridnum=null,channel='Local/s@auth-00000023;2',systemHashcode=27029873]

[2014-01-17 21:06:41,713 ERROR org.asteriskjava.live.internal.ChannelManager]
- Ignored DialEvent for unknown destination channel null with unique id null

I think this is not a big issue.
But I feel uneasy about why org.asteriskjava.live.internal.ChannelManager log ERROR.

Statistics : Posted by logankim • on Mon Jan 20, 2014 7:14 pm • Replies 1 • Views 38
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need voip and asterisk engineer

January 5, 2014, 6:06 am
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Hello,

Greetings from QLogy !!

Just wished to know if you still had requirement for this project, so that we could help you out.

We are a focused Open Source Solutions company with in-depth expertise on various Open Source solutions like Asterisk, Magento, vTiger, SugarCRM, Moodle,HRM etc.

We have been working on Asterisx & vTiger CRM since long time and have very good understanding of the same.

We would be glad to carry out the project further with you.You can send an email to me at Shilpa_Sandilya@QLogy.com.

Thanks & Regards,
Shilpa Sandilya
Skype : mishra.s417

Statistics : Posted by pcswireless • on Sun Jan 05, 2014 8:06 am • Replies 2 • Views 232
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need voip and asterisk engineer

January 5, 2014, 6:06 am
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There is a way to integrate asterisk with pretty much everything.
The short answer is yes absolutely ..

Controlling asterisk via the web is quite easy.
I would like to hear more details.

I develop remotely for several clients with similar needs.
PM me if you are still looking for help on this project.

Statistics : Posted by pcswireless • on Sun Jan 05, 2014 8:06 am • Replies 2 • Views 232
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need voip and asterisk engineer

January 5, 2014, 6:06 am
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i need to be able to control all my goip gateway ports through a webpage
like i need to be able to set priority , set specific amount of minutes on each port
instead of using the switch to control the simcards there is a way to integrate asterisk with goip gateway

please add me on skype pcs.wireless

work can be done remotely and im looking for someone for good not temporarily

i will pay over fair salary

maybe this link can help

Statistics : Posted by pcswireless • on Sun Jan 05, 2014 8:06 am • Replies 2 • Views 232
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Call hangs up due to retransmission time-out

January 20, 2014, 7:51 am
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In my rtp.conf file I've specified a stun server(forgot I had changed that so I'll post it now)

Code: [general]
rtpstart=10000
rtpend=20000
icesupport=true
stunaddr=stun.l.google.com:19302


Statistics : Posted by apwriis • on Mon Jan 20, 2014 9:51 am • Replies 2 • Views 49
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Call hangs up due to retransmission time-out

January 20, 2014, 7:51 am
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You are not configured to access systems outside your NAT. You need to specify the external address or a STUN server.

Statistics : Posted by apwriis • on Mon Jan 20, 2014 9:51 am • Replies 2 • Views 49
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Call hangs up due to retransmission time-out

January 20, 2014, 7:51 am
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Hi everyone,

I'm trying to get get a call working from a softphone with an external sip account (express talk, sisp2sip.info) to sipml5 with a sip account that is registered to my server, so far I can get the call connecting and audio flowing but, on calls to sipml, the call hangs up after 32 seconds (the call is fine from sipml to express talk though) I've checked the traces and it seems to be a retransmission error however after goggling the issue I've been unable to fix it my self my config files are:

sip.conf:
Code: [general]
udpbindaddr=0.0.0.0:5060
tcpenable=0
realm=*serverIPAddress*
transport=udp,ws,wss
;codecs
disallow=all
allow=ulaw
allow=alaw
allow=gsm
nat=force_rport,comedia

[webRTCcnct](!)
type=friend
context=from-internal
host=dynamic
icesupport=yes
secret=****
encryption=yes
avpf=yes

[1060](webRTCcnct)
defaultuser=1060
dial=SIP/1060

[1061](webRTCcncT)
defaultuser=1061
dial=SIP/1061


Extensions.conf
Code: [globals]
FIRST=SIP/1060
SECOND=SIP/1061
[unauthenticated]
exten => 1060,1,Dial(${FIRST},20)

exten => 1061,1,Dial(${SECOND},20)

exten => 200,1,Answer()
        same => n,Playback(hello-world)
        ;same => n,Hangup()

[from-internal]
exten => 1060,1,Dial(${FIRST},20)

exten => 1061,1,Dial(${SECOND},20)

exten => 200,1,Answer()
        same => n,Playback(hello-world)
        same => n,Hangup()




http.conf:
Code: [general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088


My trace from sipml5

My trace from asterisk with sip debug on

Does anyone have any Idea how to solve this? any help would be greatly appreciated

Statistics : Posted by apwriis • on Mon Jan 20, 2014 9:51 am • Replies 2 • Views 49
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NEED URGENT HELP : HELP HELP FOR STARTING SWITCH

January 22, 2014, 8:40 am
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You need to use the Biz and Jobs forum and pay normal consultancy rates.

Statistics : Posted by manibutt • on Wed Jan 22, 2014 10:40 am • Replies 1 • Views 24
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NEED URGENT HELP : HELP HELP FOR STARTING SWITCH

January 22, 2014, 8:40 am
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DEAR Friends Butt here. I want to start my own bandwidth call terminations business which works on Ip to ip calls terminations. Linux based . means i need to run calls on gsm gateway to Cell/landline with G729. on Sip. i want to start SIP company only providing solutions for Grey route. Please explain me every thing what i need in the starting & how much does it cost me.? For Exp: http://www.syncswitch.com OR http://www.appleswitch.us. i need tto provide these types services they run services on Linux Puppy with ASerisk SVN -branch -1.8 -r 3320213M. please only reply me if you know all these. your & mine time is important to do not laugh on my post if any Expert here. so i am the new & doing call termination business so treat me a new comer. so i will be very thankful to you all if any one help me for this.

Statistics : Posted by manibutt • on Wed Jan 22, 2014 10:40 am • Replies 1 • Views 24
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Attended transfer as default on refer

January 22, 2014, 7:35 am
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Asterisk supports REFER/Replaces. I've no idea about your soft phone. Any SIP hard phone would.

Statistics : Posted by iorlas • on Wed Jan 22, 2014 9:35 am • Replies 1 • Views 23
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Attended transfer as default on refer

January 22, 2014, 7:35 am
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I'm using MicroSIP and SFLphone as SIP softphones. Both has ability to transfer calls with special button.
When I enter exten and press it, softphone sends REFER udp package and then hungs up - so, it is blind transfer.
I want to use attended transfer instead. Is it possible?

Statistics : Posted by iorlas • on Wed Jan 22, 2014 9:35 am • Replies 1 • Views 23
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XMPP with Asterisk

January 22, 2014, 1:14 am
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For step by step instructions, please hire a consultant on the Biz and Jobs forum.

XMPP is a protocol so is incapable of accessing a database.

Statistics : Posted by musman87 • on Wed Jan 22, 2014 3:14 am • Replies 1 • Views 28
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XMPP with Asterisk

January 22, 2014, 1:14 am
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Hi,
I want to configure Asterisk with XMPP, can any body please help me finding a step by step guideline for that. Also is it possible that both XMPP and Asterisk use the same database.

Thanks in advance

Statistics : Posted by musman87 • on Wed Jan 22, 2014 3:14 am • Replies 1 • Views 28
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Allow users to send files

January 22, 2014, 1:23 am
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You will need to do that outside of the telephone system.

Statistics : Posted by musman87 • on Wed Jan 22, 2014 3:23 am • Replies 3 • Views 49
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Allow users to send files

January 22, 2014, 1:23 am
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I donot want them to email, rather they can transfer file directly between each other.

Statistics : Posted by musman87 • on Wed Jan 22, 2014 3:23 am • Replies 3 • Views 49
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Allow users to send files

January 22, 2014, 1:23 am
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Email server?

Statistics : Posted by musman87 • on Wed Jan 22, 2014 3:23 am • Replies 3 • Views 49
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Allow users to send files

January 22, 2014, 1:23 am
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Hi,
I have configured an Asterisk server and its working perfectly, now I want my users to send files to each other, in case if the recipient is offline, server should save the file and forwards it to recipient once they are online.
Can anybody suggest me which external service i can use for this purpose.. Thanks in Advance.

Statistics : Posted by musman87 • on Wed Jan 22, 2014 3:23 am • Replies 3 • Views 49
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