Re: [asterisk-users] Asterisk and Amazon Web Services
On Tue, 27 Jul 2010, Kyle Kienapfel wrote: On Tue, Jul 27, 2010 at 12:50 PM, Roderick A. Anderson raand...@cyber-office.net wrote: Anyone tried installing Asterisk in a AWS server? It probably works as well as it does virtualized other ways. I've seen peoples opinions on how virtualizing asterisk is a bad idea and might have trouble related to timing and hosting conferences. I am successfully running asterisk in a virtualised environment - sort of, it's LXC - one kernel, multiple containers. MeetMe demos worked OK, although none of the current instances use it in anger. I have full control over the host servers, so I can balance the number of instances over physical servers better than just picking random server... Gordon -- _ -- 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] Random DTMF Tones Only on heard on ATA
Travis Langhals tra...@netitek.com writes: [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin '1' received on SIP/5211-0078 Is SIP/5211 a Linksys or a Grandstream or something else? Do you have relaxdtmf=no? Also, your Asterisk version numbers are incorrect. Do you mean 1.6.2.10? /Benny -- _ -- 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] Asterisk and Amazon Web Services
Randy R randulo2...@gmail.com writes: I'd think twice about trying this, taking into account the recent spate of attacks to so many of us coming from Amazon EC2 and particularly their answer to complaints, which was something like Deal with it. Indeed, my personal threshold for dealing with EC2 traffic has become if in doubt, ban it. /Benny -- _ -- 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] Urgent help = RUBY AGI
Update on this - breaktrough! :-)
Finally, I was able to do it. Yes you were right again as you said, I saw
that you mentioned using macros but for some reason I thought that macros
were not available in asterisk 1.2...
So what I managed to do is o start a macro and get info about the channel
that picked up the call.
Also, as said in the thread, i was able to start moh beside the macro and it
all works like charm.
The only thing that I cant do so far, is capturing the ${CHANNEL} variable
in the ruby script that started the macro.
Is that variable accessible from the ruby script too or just from the macro?
Here's a snippet from my ruby script:
dial_params ||M(testing)m(moh-0900...@moh_id})
if moh_available?()
1.times do
r = $agi.exec('DIAL', dial_params)
r = $agi.get_variable('DIALSTATUS')
retry if r.message.include?('BUSY')
end
and further below:
$loc = testing
$my.query(UPDATE call_log SET
local='#{$loc}', endtime = NOW() WHERE id = #{call_log_id})
Works fine, but as soon as I try:
$loc = ${CHANNEL} or something like that - it breaks. Any idea how to pass
that ${CHANNEL} to my ruby script and use it to update DB in that query?
Thanks a bunch !
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Tuesday, July 27, 2010 5:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Urgent help = RUBY AGI
I have never used 1.2.9.1 or anything in the 1.2.x range so I can not give
you an exact solution but I can tell you that the script that you are using
will not work. In the dial command you need to add the M option which will
call a macro when the call is connected. In that macro you can then find the
channel that answered the call and do what you want from there. You can call
another AGI or set variables or whatever. If agi.exec works like a dialplan
step then the dial step will hang if the call is answered and the
agi.get_variable statement will not execute unless the call was not
answered.
Try
r = $agi.exec('DIAL', SIP/voipuserZap/32Zap/33Zap/34Zap/35,,M(testing))
And then have something like this in extensions.conf
[macro-testing]
exten = s,1,DumpChan()
You will see that this macro runs when the call is answered and you will see
on the CLI all the variables that are available to you. ${CHANNEL} will have
SIP/ voipuser-e989 in your example below.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 7:21 AM, Zarko Zivanovic wrote:
Here's something that should be easy for RUBY pro's.
Here is a script:
1.times do
r = $agi.exec('DIAL',
SIP/voipuserZap/32Zap/33Zap/34Zap/35)
r = $agi.get_variable('DIALSTATUS')
# $agi.set_variable(' WHOANSWERED ',...)
retry if r.message.include?('BUSY')
end
when it's executed it shows this in the console:
AGI Rx ANSWER
AGI Tx 200 result=0
AGI Rx EXEC DIAL SIP/voipuserZap/32Zap/33Zap/34Zap/35
-- AGI Script Executing Application: (DIAL) Options:
(SIP/voipuserZap/32Zap/33Zap/34Zap/35)
-- Called voipuser
-- Called 32
-- Called 33
-- Called 34
-- Called 35
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/voipuser-e989 is ringing
-- SIP/ voipuser-e989 answered Zap/1-1
What we need is to be able to populate the variable WHOANSWERED with info
SIP/ voipuser
In this case, or whoever answers next time.
Thanks in advance!
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New
[asterisk-users] Redirecting a call to another extension using asterisk java
Hi, My problem is as follows. I registered an xlite client and dialed 1500 extension. In the extensions.conf i set as follows. exten=1500,1,AGI(localhost// hello.agi. This hello.agi when connected plays a greeting message. Once this is connected from the script i want to transfer the call to another extension say 1600. How do i achieve this. I tried using ChannelRedirect but it didnt work. I want this transfer to happen from the script. Any suggestions please? Thanks in Advance, Jahnavi. -- _ -- 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] Urgent help = RUBY AGI
Just to mention... I also tried:
$my.query(UPDATE call_log SET local='#{$CHANNEL}', endtime = NOW() WHERE id
= #{call_log_id})
But then the local is empty - meaning $CHANNEL is empty in ruby.
The question is how do i pass that macro dumpchan data to ruby?
Zrko
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Tuesday, July 27, 2010 6:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Urgent help = RUBY AGI
You can put multiple options in the dial command if that is what you are
asking.
And by the way several emails, including a previous one of mine, told you to
use the M option and a macro.
In this email I gave you more detailed information but if you had done core
show application dial on CLI you should have been able to ask more directed
questions.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 9:28 AM, Zarko Zivanovic wrote:
Jim thanks.
I will test this first thing in the morning as I am out of the office now.
As a matter of fact I cant wait to test this, as it has been the first
reasonable thing that looks like it could work.
In the meantime , do you happen to know if there is a way to call both macro
(M) and music on hold (m) in that $agi.exec line?
or is the right thing to do to place moh command in macro?
As I said, I cant wait to try it first thing in the morning and tell you
(and others) how it went. I am sure this will be the good reference to other
people looking for the same thing online as I have found quite a bunch of
similar open threads.
Zarko
On Tue, Jul 27, 2010 at 5:31 PM, Jim Dickenson dicken...@cfmc.com wrote:
I have never used 1.2.9.1 or anything in the 1.2.x range so I can not give
you an exact solution but I can tell you that the script that you are using
will not work. In the dial command you need to add the M option which will
call a macro when the call is connected. In that macro you can then find the
channel that answered the call and do what you want from there. You can call
another AGI or set variables or whatever. If agi.exec works like a dialplan
step then the dial step will hang if the call is answered and the
agi.get_variable statement will not execute unless the call was not
answered.
Try
r = $agi.exec('DIAL', SIP/voipuserZap/32Zap/33Zap/34Zap/35,,M(testing))
And then have something like this in extensions.conf
[macro-testing]
exten = s,1,DumpChan()
You will see that this macro runs when the call is answered and you will see
on the CLI all the variables that are available to you. ${CHANNEL} will have
SIP/ voipuser-e989 in your example below.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 7:21 AM, Zarko Zivanovic wrote:
Here's something that should be easy for RUBY pro's.
Here is a script:
1.times do
r = $agi.exec('DIAL',
SIP/voipuserZap/32Zap/33Zap/34Zap/35)
r = $agi.get_variable('DIALSTATUS')
# $agi.set_variable(' WHOANSWERED ',...)
retry if r.message.include?('BUSY')
end
when it's executed it shows this in the console:
AGI Rx ANSWER
AGI Tx 200 result=0
AGI Rx EXEC DIAL SIP/voipuserZap/32Zap/33Zap/34Zap/35
-- AGI Script Executing Application: (DIAL) Options:
(SIP/voipuserZap/32Zap/33Zap/34Zap/35)
-- Called voipuser
-- Called 32
-- Called 33
-- Called 34
-- Called 35
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/voipuser-e989 is ringing
-- SIP/ voipuser-e989 answered Zap/1-1
What we need is to be able to populate the variable WHOANSWERED with info
SIP/ voipuser
In this case, or whoever answers next time.
Thanks in advance!
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database 5317 (20100727) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com http://www.eset.com/
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asterisk-users mailing list
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[asterisk-users] Answered call not bridged
Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a customer complaining about the issue [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, default|xxx|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (default,02034684373,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, enge-xx|s|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (enge-02034684373,s,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing NoOp(SIP/PACK501-480b08c0, ) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Wait(SIP/PACK501-480b08c0, 2) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Set(SIP/PACK501-480b08c0, CALLERID(num)=PACK501) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Dial(SIP/PACK501-480b08c0, SIP/ENGE103|20) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Called ENGE103 [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing *** AT this point the customer had answered and I was talking to him!! [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Nobody picked up in 2 ms [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Executing Voicemail(SIP/PACK501-480b08c0, 1...@enge-local|u) [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-theperson' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/1' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/0' (language 'en') [2010-07-28 11:07:51] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/3' (language 'en') [2010-07-28 11:07:52] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-isunavail' (language 'en') [2010-07-28 11:07:53] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-intro' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'beep' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- Recording the message [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=0, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav49, 0xb75e60 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=1, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: gsm, 0xb20720 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=2, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav, 0xa1c850 [2010-07-28 11:08:00] VERBOSE[6554] logger.c: -- User hung up [2010-07-28 11:08:00] VERBOSE[6554] logger.c: == Spawn extension (enge-02034684373, s, 5) exited non-zero on 'SIP/PACK501-480b08c0' The customer is using Aastra phones but it's happened once with us when I was using a Snom phone. I'm trying to consistently replicate the issue so that I can analyse it properly but have not been able to so far. Has anyone ever experienced anything like this? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Answered call not bridged
If you run a sip debug at the same time you will get some more usefull logs. What sip client are you using? Ishfaq Malik wrote: Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a customer complaining about the issue [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, default|xxx|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (default,02034684373,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, enge-xx|s|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (enge-02034684373,s,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing NoOp(SIP/PACK501-480b08c0, ) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Wait(SIP/PACK501-480b08c0, 2) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Set(SIP/PACK501-480b08c0, CALLERID(num)=PACK501) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Dial(SIP/PACK501-480b08c0, SIP/ENGE103|20) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Called ENGE103 [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing *** AT this point the customer had answered and I was talking to him!! [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Nobody picked up in 2 ms [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Executing Voicemail(SIP/PACK501-480b08c0, 1...@enge-local|u) [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-theperson' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/1' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/0' (language 'en') [2010-07-28 11:07:51] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/3' (language 'en') [2010-07-28 11:07:52] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-isunavail' (language 'en') [2010-07-28 11:07:53] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-intro' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'beep' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- Recording the message [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=0, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav49, 0xb75e60 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=1, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: gsm, 0xb20720 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=2, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav, 0xa1c850 [2010-07-28 11:08:00] VERBOSE[6554] logger.c: -- User hung up [2010-07-28 11:08:00] VERBOSE[6554] logger.c: == Spawn extension (enge-02034684373, s, 5) exited non-zero on 'SIP/PACK501-480b08c0' The customer is using Aastra phones but it's happened once with us when I was using a Snom phone. I'm trying to consistently replicate the issue so that I can analyse it properly but have not been able to so far. Has anyone ever experienced anything like this? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Answered call not bridged
On receiving a call, try using the 'Answer()' command before anything else. I once had some issues, though not similar, which were solved by this command, as it sends back a SIP acknowledgement to the calling party which is otherwise not sent. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-28 6:30 AM, Ishfaq Malik i...@pack-net.co.uk wrote: Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a customer complaining about the issue [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, default|xxx|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (default,02034684373,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, enge-xx|s|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (enge-02034684373,s,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing NoOp(SIP/PACK501-480b08c0, ) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Wait(SIP/PACK501-480b08c0, 2) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Set(SIP/PACK501-480b08c0, CALLERID(num)=PACK501) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Dial(SIP/PACK501-480b08c0, SIP/ENGE103|20) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Called ENGE103 [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing *** AT this point the customer had answered and I was talking to him!! [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Nobody picked up in 2 ms [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Executing Voicemail(SIP/PACK501-480b08c0, 1...@enge-local|u) [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-theperson' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/1' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/0' (language 'en') [2010-07-28 11:07:51] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/3' (language 'en') [2010-07-28 11:07:52] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-isunavail' (language 'en') [2010-07-28 11:07:53] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-intro' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'beep' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- Recording the message [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=0, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav49, 0xb75e60 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=1, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: gsm, 0xb20720 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=2, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav, 0xa1c850 [2010-07-28 11:08:00] VERBOSE[6554] logger.c: -- User hung up [2010-07-28 11:08:00] VERBOSE[6554] logger.c: == Spawn extension (enge-02034684373, s, 5) exited non-zero on 'SIP/PACK501-480b08c0' The customer is using Aastra phones but it's happened once with us when I was using a Snom phone. I'm trying to consistently replicate the issue so that I can analyse it properly but have not been able to so far. Has anyone ever experienced anything like this? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Answered call not bridged
Hi The problem is that this is a production server with usually about 10 concurrent calls going on and also that if I just run a sip debug on the customers peer, I still don't know when it's either this issue or if it genuinely went to voicemail. That's why I'm trying to consistently replicate the issue so I can do a controlled sip debug on it :( They are using Aastra 51i/2.1.0.2145 Thanks Ish On 28/07/10 11:47, Gareth Blades wrote: If you run a sip debug at the same time you will get some more usefull logs. What sip client are you using? Ishfaq Malik wrote: Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a customer complaining about the issue [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, default|xxx|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (default,02034684373,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, enge-xx|s|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (enge-02034684373,s,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing NoOp(SIP/PACK501-480b08c0, ) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Wait(SIP/PACK501-480b08c0, 2) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Set(SIP/PACK501-480b08c0, CALLERID(num)=PACK501) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Dial(SIP/PACK501-480b08c0, SIP/ENGE103|20) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Called ENGE103 [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing *** AT this point the customer had answered and I was talking to him!! [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Nobody picked up in 2 ms [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Executing Voicemail(SIP/PACK501-480b08c0, 1...@enge-local|u) [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-theperson' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/1' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/0' (language 'en') [2010-07-28 11:07:51] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/3' (language 'en') [2010-07-28 11:07:52] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-isunavail' (language 'en') [2010-07-28 11:07:53] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-intro' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'beep' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- Recording the message [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=0, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav49, 0xb75e60 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=1, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: gsm, 0xb20720 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=2, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav, 0xa1c850 [2010-07-28 11:08:00] VERBOSE[6554] logger.c: -- User hung up [2010-07-28 11:08:00] VERBOSE[6554] logger.c: == Spawn extension (enge-02034684373, s, 5) exited non-zero on 'SIP/PACK501-480b08c0' The customer is using Aastra phones but it's happened once with us when I was using a Snom phone. I'm trying to consistently replicate the issue so that I can analyse it properly but have not been able to so far. Has anyone ever experienced anything like this? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Answered call not bridged
Hi Unfortunately this isn't an option as we allow customers to forward incoming calls back out to POTS or mobile. If we use an explicit Answer() all forwarded calls show as answered even if they weren't by the POTS or mobile end point. Ish On 28/07/10 11:48, Zeeshan Zakaria wrote: On receiving a call, try using the 'Answer()' command before anything else. I once had some issues, though not similar, which were solved by this command, as it sends back a SIP acknowledgement to the calling party which is otherwise not sent. Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com On 2010-07-28 6:30 AM, Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote: Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a customer complaining about the issue [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, default|xxx|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (default,02034684373,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Goto(SIP/PACK501-480b08c0, enge-xx|s|1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Goto (enge-02034684373,s,1) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing NoOp(SIP/PACK501-480b08c0, ) [2010-07-28 11:07:25] VERBOSE[6554] logger.c: -- Executing Wait(SIP/PACK501-480b08c0, 2) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Set(SIP/PACK501-480b08c0, CALLERID(num)=PACK501) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Executing Dial(SIP/PACK501-480b08c0, SIP/ENGE103|20) [2010-07-28 11:07:27] VERBOSE[6554] logger.c: -- Called ENGE103 [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing *** AT this point the customer had answered and I was talking to him!! [2010-07-28 11:07:28] VERBOSE[6554] logger.c: -- SIP/ENGE103-009140e0 is ringing [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Nobody picked up in 2 ms [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- Executing Voicemail(SIP/PACK501-480b08c0, 1...@enge-local|u) [2010-07-28 11:07:48] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-theperson' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/1' (language 'en') [2010-07-28 11:07:50] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/0' (language 'en') [2010-07-28 11:07:51] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'digits/3' (language 'en') [2010-07-28 11:07:52] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-isunavail' (language 'en') [2010-07-28 11:07:53] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'vm-intro' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- SIP/PACK501-480b08c0 Playing 'beep' (language 'en') [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- Recording the message [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=0, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav49, 0xb75e60 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=1, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: gsm, 0xb20720 [2010-07-28 11:07:59] VERBOSE[6554] logger.c: -- x=2, open writing: /var/spool/asterisk/voicemail/enge-local/103/tmp/S85HqQ format: wav, 0xa1c850 [2010-07-28 11:08:00] VERBOSE[6554] logger.c: -- User hung up [2010-07-28 11:08:00] VERBOSE[6554] logger.c: == Spawn extension (enge-02034684373, s, 5) exited non-zero on 'SIP/PACK501-480b08c0' The customer is using Aastra phones but it's happened once with us when I was using a Snom phone. I'm trying to consistently replicate the issue so that I can analyse it properly but have not been able to so far. Has anyone ever experienced anything like this? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] Grab voicemail WAV file when done
On 10-07-27 10:02 PM, Michelle Dupuis wrote:
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
[leif.mad...@asteriskdocs.org]
Sent: Tuesday, July 27, 2010 9:22 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Grab voicemail WAV file when done
On 10-07-27 08:38 PM, Michelle Dupuis wrote:
I need to grab the voicemail WAV file once the voicemail command is done.
Is there a hook to be notified that voicemail is done, and get the name of
the recorded file?
Look at the 'externnotify' option to voicemail.conf.
The problem is that I need to catch the filename in the dialplan, since I
will be recording several other files and concatenating them with SOX.
Per doc/tex/channelvariables.tex (in 1.8.0-beta1 at least)
\subsection{The VoiceMail() application}
\begin{verbatim}
${VM_CATEGORY} Sets voicemail category
${VM_NAME}* Full name in voicemail
${VM_DUR} * Voicemail duration
${VM_MSGNUM} * Number of voicemail message in mailbox
${VM_CALLERID}* Voicemail Caller ID (Person leaving vm)
${VM_CIDNAME} * Voicemail Caller ID Name
${VM_CIDNUM} * Voicemail Caller ID Number
${VM_DATE}* Voicemail Date
${VM_MESSAGEFILE} * Path to message left by caller
\end{verbatim}
(Thanks Russell!)
Leif Madsen.
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[asterisk-users] IAX authentication oddity - Known issue? Fixed?
Hi, I had the following odd behaviour in Asterisk 1.2 - We are migrating to 1.6, and I will re-test ASAP, though it is quite hard to replicate, but I am curious to know whether it is a known IAX issue in 1.2. We had 2 users in iax.conf: [user1] username=user1 secret=secret1 context=context1 host=iax.hostname.com [user2] username=user2 secret= context=context2 host=dynamic deny=0.0.0.0/0.0.0.0 allow=1.2.3.0/255.255.255.0 A call came in with username=user2, the call was from the valid IP range specified in [user2], and the IAX debug trace showed the call as UNAUTHENTICATED. So far so good. The issue is that once the call was in, the channel-name was allocated as IAX/user1-xxx (instead of IAX/user2-xxx) and the call jumped to context1 instead of context2. I believe that the source IP address for the call DOES fall into the list of IP addresses that resolve using iax.hostname.com. I am concerned! Regards, Steve -- _ -- 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] Passing Variables From Dial Macro To Parent Ruby
Thanks to help from Jim Dickenson I managed to start a macro and get info
about the channel that picked up the call from my ruby script.
The only thing that I cant do so far, is capturing the ${CHANNEL} variable
in the ruby script that started the macro.
Is that variable accessible from the ruby script too or just from the macro?
Here's a snippet from my ruby script:
dial_params ||M(testing)m(moh-0900...@moh_id})
if moh_available?()
1.times do
r = $agi.exec('DIAL', dial_params)
r = $agi.get_variable('DIALSTATUS')
retry if r.message.include?('BUSY')
end
and further below:
$loc = testing
$my.query(UPDATE call_log SET
local='#{$loc}', endtime = NOW() WHERE id = #{call_log_id})
Works fine, but as soon as I try:
$loc = ${CHANNEL} or something like that - it breaks. Any idea how to pass
that ${CHANNEL} to my ruby script and use it to update DB in that query?
Thanks a bunch !
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Re: [asterisk-users] Answered call not bridged
Hi! - upgrade to a current 1.4 version, 1.4.17 is very old (you probably run this because of the zaptel -- dahdi change, but still) - do you have a SIP proxy or any SIP-aware hardware in your network that might play tricks on you, e.g. a SIP ALG (application layer gateway) on your Internet router or something similar? - enable SIP debugging on your phone and check its logs; you could also do a packet capture on your router to see what exactly is happening and if Asterisk is somehow being cut out of the loop - see if canreinvite=no somehow helps; disable STUN on your phone inside the LAN, and maybe even block direct Internet traffic for your LAN phones so that they must go through Asterisk. Philipp -- _ -- 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] Urgent help = RUBY AGI
Works like charm Danny,
Tested and works fine.
I sent this to Jim, but this is something you could know:
I managed to start a macro and get info about the channel that picked up the
call.
Also, as said in the thread, i was able to start moh beside the macro and it
all works like charm.
The only thing that I cant do so far, is capturing the ${CHANNEL} variable
in the ruby script that started the macro.
Is that variable accessible from the ruby script too or just from the macro?
Here's a snippet from my ruby script:
dial_params ||M(testing)m(moh-0900...@moh_id})
if moh_available?()
1.times do
r = $agi.exec('DIAL', dial_params)
r = $agi.get_variable('DIALSTATUS')
retry if r.message.include?('BUSY')
end
and further below:
$loc = testing
$my.query(UPDATE call_log SET
local='#{$loc}', endtime = NOW() WHERE id = #{call_log_id})
Works fine, but as soon as I try:
$loc = ${CHANNEL} or something like that - it breaks. Any idea how to pass
that ${CHANNEL} to my ruby script and use it to update DB in that query?
Thanks a bunch !
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, July 27, 2010 6:32 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Urgent help = RUBY AGI
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zarko
Zivanovic
Subject: Re: [asterisk-users] Urgent help = RUBY AGI
snip
In the meantime , do you happen to know if there is a way to call both macro
(M) and music on hold (m) in that $agi.exec line?
or is the right thing to do to place moh command in macro?
This should work:
r = $agi.exec('DIAL', SIP/voipuserZap/32Zap/33Zap/34Zap/35,,mM(testing))
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Re: [asterisk-users] Passing Variables From Dial Macro To Parent Ruby
Hi! Three notes: * as others have already mentioned: personally I would not Dial() from within AGI using EXEC, but rather set extension and context and then let the dialplan handle the Dial, and therefore complete that AGI before the Dial; then possibly run another AGI after the call in the h extension (even if that might not scale so well it is usually just fine). * the second call leg is already gone when call control returns to your AGI, that is why you cannot read its variables. The only way is to use your M() Macro to store the value you are interested in, either by putting it into the CDR(userfield), or by using the SHARED() function (there is a backport for Asterisk 1.4) that can export the data to the originating channel. * you need to do a GET VARIABLE in your AGI, not sure if this works flawlessly with the CHANNEL variables though Philipp -- _ -- 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] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
Leif Madsen wrote: I have a client using QueueMetrics and they seem to be fairly pleased with it. Their response times on issues has been pretty good from what I can tell (I had the client communicate with them directly where necessary). Unless you build it yourself, I'm not sure there is any good + free queue metrics program. Queue's typically are a money generating adventure and as such makes sense for this type of application to be a pay-for system. Bruce, I'll throw in another vote for QueueMetrics. It's not free, but you get a lot of value for your money. The application is solid and the developer is top notch. Check out a demo at: http://queuemetrics.com/demosys.jsp Or get a get a free temporary (30-day) license at: http://queuemetrics.com/sendDemoLicence.jsp Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] FreeTDS (Microsoft MsSQL 2008) and CDR
I resolved this isue using odbc. On Mon, Jul 26, 2010 at 11:27 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, Jul 26, 2010 at 10:05:27AM +0200, Andraž wrote: Hi, I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from sources. I installed freetds-common,freetds-dev, libct4, libsybdb5, freetds-bin, but, when I run configure and then make menuconfig in section Call Detail Recording - cdr_tds it's disabled. It only writes that Depends on: freetds(E). On another server (same configuration) I installed the same packages, and it's working fine. Any suggestions, what I did wrong? Have you re-ron ./configure #? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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
[asterisk-users] AMI Monitor - one file
Hi, I try to record a conversation trough AMI (Monitor). In the documentation is written, that if you would like recordings to be in one file, just use Mix: 1. I use this parameter, but still I have 2 files. Filename-in.wav and Filename-out.wav. Regards Andraž -- _ -- 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] fail2ban and pf
Hi, Since f2b is one of the topics du jour here, I was wondering if someone would mind telling me what these pf stats mean: Evaluations: 964303 Packets: 12176 Bytes: 648408 States: 0 Looks like pf examined nearly a million cases from fail2ban in 24h? thanks, /r -- _ -- 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] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
On Tue, Jul 27, 2010 at 6:08 PM, bruce bruce bruceb...@gmail.com wrote: :-) I knew someone would bring up FreePBX. I have FreePBX installed and it's not good for Queues at all. It's using the reporting tool from Areski and One of the several things you asked for was GUI for cdr database logs. FreePBX is good for putting a gui on top of doing database cdr log searching. This of course, assumes your colleagues know enough about using a database, to be able to use a pull-down web form to do filtering, and also know enough to not keep clicking when they ask for such a broad query that it takes several seconds for the database to return the results. -- _ -- 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] 1 second Audio Lag
Hi! I'm working for Zoiper, you can contact us directly on supp...@zoiper.com Zoa I will do a test call from a soft phone to my mobile. I can speak into my headset and the audio is heard instantly. But if I speak into my mobile there is a 1-2 second delay in the Audio. I am using SIP. I am only finding it in the Zoiper Softphones that we are using. We are able to make a call without lag on the X-lite softphone no problem. This looks very similar: https://issues.asterisk.org/view.php?id=17404 0017404: [regression] audio delay when bridging calls related to timestamp mismatch when answering an inbound call, the remote party hears a delay from 1-3 seconds. The audio is being transmitted, but the rtp timestamps take a huge jump when the call is answered even though the rtp sequencing is correct. This started occurring after 1.4.28. reproduced with 1.4.30, 1.4.32 and SVN from 05/25/2010. related to 0017007: [patch] RTP Timestamp changes after transfer, but SSRC not and the markerbit ist not set (one-way audio after transfer) Philipp -- _ -- 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] Subscribe Problem - Zombie Channel
Hi List, Asterisk 1.4.22 built by root @ carl on a i686 Purely SIP Linksys SPA962 with 932 sidecar and also Cisco SPA508 / 525G with Sidecars Have an issue with this happening with a number of my customers. Customer hits the ringing BLF on the sidecar to pickup the call incoming on another handset. Not always but enough to cause me a problem it leaves a channel open. At the moment, this is the output from core show channels on one of the affected sites: core show channels Channel Location State Application(Data) SIP/102--08e1 *...@from-inside Down(None) SIP/102--08d6 *...@from-inside Ring(None) SIP/102--08d7 *...@from-inside Ring(None) 3 active channels 0 active calls I can't get rid of them, if I: soft hangup SIP/102--08e149c0 Requested Hangup on channel 'SIP/102--08e149c0' and thats all that happens. core show channels Channel Location State Application(Data) SIP/102--08e1 *...@from-inside Down(None) SIP/102--08d6 *...@from-inside Ring(None) SIP/102--08d7 *...@from-inside Ring(None) 3 active channels 0 active calls The only way to free them up is to force a restart. restart now Any clues on how I can debug this and try to sort it or even if anyone has come across this. Many thanks in advance. Brian -- _ -- 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] fail2ban and pf
On Wed, Jul 28, 2010 at 6:38 AM, Randy R randulo2...@gmail.com wrote: Hi, Since f2b is one of the topics du jour here, I was wondering if someone would mind telling me what these pf stats mean: Evaluations: 964303 Packets: 12176 Bytes: 648408 States: 0 Looks like pf examined nearly a million cases from fail2ban in 24h? thanks, /r -- _ -- 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 80 or so blocked hosts? 964303/12176=~79.19 evaluations being more than packets looks like its going through multiple rules to think about packets. White listing your itsp and other traffic you know you like; before the list of banned computers might reduce that a bit. -- _ -- 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] Redirecting a call to another extension using asterisk java
On Wed, Jul 28, 2010 at 02:59:23PM +0530, Janu Mukherjee wrote: Hi, My problem is as follows. I registered an xlite client and dialed 1500 extension. In the extensions.conf i set as follows. exten=1500,1,AGI(localhost// hello.agi. I wonder why you use the odd name 'localhost' here. This hello.agi when connected plays a greeting message. Once this is connected from the script i want to transfer the call to another extension say 1600. How do i achieve this. I tried using ChannelRedirect but it didnt work. What you write here is what didn't happen. Can you give more details of what did happen? E.g. a CLI trace of when you tried to trasfer using that method? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Nat issue one way audio on IP dial
hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can hear the called user but the called user can not here the caller voice. If the caller calls the other user by username instead of IP:Port , the voice is perfect both ways. what i have noticed is that IP:Port dial is missing a parameter rinstance in Contact , To headers for adf. what is rinstance for? Also something with Contact header seems fishy. or RTP issue. that is Dial(SIP/adf,30,r) works fine with bothway audio, but Dial(SIP/116.18.35.235:28614,30,r) has one way audio. / \ | | this is IP:Port of of adf please help as it's almost 2 weeks and i have found to suitable answer from any forum. I nead to know what can i do to modify Headers or settings in conf files to correct this problem. Below is the conf of calling user [pepsi] username=pepsi type=friend secret=123456 qualify=yes nat=no insecure=port,invite incominglimit=1 outgoinglimit=1 host=dynamic dtmfmode=rfc2833 context=out canreinvite=yes callerid=pepsi coke 12345678901 accountcode=6:0:pepsi amaflags=default disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm Below is the conf of called user [adf] username=adf type=friend secret=123456 qualify=yes nat=yes insecure=port,invite incominglimit=2 outgoinglimit=2 host=dynamic dtmfmode=rfc2833 context=user canreinvite=yes callerid=adf xyz 11223344556 accountcode=1:0:adf amaflags=default disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm below is my sip debug after dialing Audio is at 79.80.x.x port 16238 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 116.18.35.235:28614: INVITE sip:a...@116.18.35.235:28614 SIP/2.0 Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport From: pepsi coke sip:12345678...@79.80.x.x:5678;tag=as42ec768c To: sip:a...@116.18.35.235:28614 Contact: sip:12345678...@79.80.x.x:5678 Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jul 2010 15:10:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 9626 9626 IN IP4 79.80.x.x s=session c=IN IP4 79.80.x.x t=0 0 m=audio 16238 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jul 21 11:10:22] WARNING[23814]: chan_sip.c:2872 sip_call: Setting auto-congest time to 15000 ms. -- Called a...@116.18.35.235:28614 ast-server*CLI --- SIP read from 116.18.35.235:28614 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678 Contact: sip:a...@116.18.35.235:28614 To: sip:a...@116.18.35.235:28614;tag=d54e632c From: pepsi cokesip:12345678...@79.80.x.x:5678;tag=as42ec768c Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x CSeq: 102 INVITE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 - --- (9 headers 0 lines) --- -- SIP/116.18.35.235:28614-007f4660 is ringing ast-server*CLI --- SIP read from 116.18.35.235:28614 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678 Contact: sip:a...@116.18.35.235:28614 To: sip:a...@116.18.35.235:28614;tag=d54e632c From: pepsi cokesip:12345678...@79.80.x.x:5678;tag=as42ec768c Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 185 v=0 o=- 6 2 IN IP4 192.168.0.12 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.0.12 t=0 0 m=audio 55246 RTP/AVP 8 0 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (11 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.12:55246 Found description format telephone-event for ID 101 Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.12:55246 list_route: hop: sip:a...@116.18.35.235:28614 [Jul 21 11:10:27] DEBUG[9707]: chan_sip.c:5695 reqprep: Strict routing enforced for session
[asterisk-users] what is rinstance parameter in sip header
hello i was wondering what is the use of rinstance in SIP Headers. I noticed that this parameter is visible only when there is NAT invloved. I am experiencing one way audio when dialing a registered user by his IP:port. I this case rinstance parameter is missing. when i dial SIP/username audio is fine but when i dial SIP/x.x.x.x:port there is one way audion. Also please tell me what can go wrong by dialing by ip:port.?? Best regards, Nasir Javaid -- _ -- 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] IAX authentication oddity - Known issue? Fixed?
On Wednesday 28 July 2010 06:49:01 Steve Davies wrote: Hi, I had the following odd behaviour in Asterisk 1.2 - We are migrating to 1.6, and I will re-test ASAP, though it is quite hard to replicate, but I am curious to know whether it is a known IAX issue in 1.2. We had 2 users in iax.conf: [user1] username=user1 secret=secret1 context=context1 host=iax.hostname.com [user2] username=user2 secret= context=context2 host=dynamic deny=0.0.0.0/0.0.0.0 allow=1.2.3.0/255.255.255.0 A call came in with username=user2, the call was from the valid IP range specified in [user2], and the IAX debug trace showed the call as UNAUTHENTICATED. So far so good. The issue is that once the call was in, the channel-name was allocated as IAX/user1-xxx (instead of IAX/user2-xxx) and the call jumped to context1 instead of context2. I believe that the source IP address for the call DOES fall into the list of IP addresses that resolve using iax.hostname.com. I don't see a 'type' argument to either of the above, so neither of these would at all be used. That said, you're assuming that the deny and allow determine who is allowed to be user2. That's incorrect. They permit what packets will even reach user2, and a registration needs to occur for the host address to become something other than 0.0.0.0 (which is the default, unless you have a defaultip parameter). Hence, user2 won't match anything at all until a registration packet comes in that passes your deny/allow ACL. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Nat issue one way audio on IP dial
Do you have your softphone setup to use a stun server so it can send it's public IP address in the SIP packets? I see in the SIP debug output a 192.168 address for the RTP packets to go to which of course will not work. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 28, 2010, at 9:23 AM, Nasir Javaid wrote: hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can hear the called user but the called user can not here the caller voice. If the caller calls the other user by username instead of IP:Port , the voice is perfect both ways. what i have noticed is that IP:Port dial is missing a parameter rinstance in Contact , To headers for adf. what is rinstance for? Also something with Contact header seems fishy. or RTP issue. that is Dial(SIP/adf,30,r) works fine with bothway audio, but Dial(SIP/116.18.35.235:28614,30,r) has one way audio. / \ | | this is IP:Port of of adf please help as it's almost 2 weeks and i have found to suitable answer from any forum. I nead to know what can i do to modify Headers or settings in conf files to correct this problem. Below is the conf of calling user [pepsi] username=pepsi type=friend secret=123456 qualify=yes nat=no insecure=port,invite incominglimit=1 outgoinglimit=1 host=dynamic dtmfmode=rfc2833 context=out canreinvite=yes callerid=pepsi coke 12345678901 accountcode=6:0:pepsi amaflags=default disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm Below is the conf of called user [adf] username=adf type=friend secret=123456 qualify=yes nat=yes insecure=port,invite incominglimit=2 outgoinglimit=2 host=dynamic dtmfmode=rfc2833 context=user canreinvite=yes callerid=adf xyz 11223344556 accountcode=1:0:adf amaflags=default disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm below is my sip debug after dialing Audio is at 79.80.x.x port 16238 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 116.18.35.235:28614: INVITE sip:a...@116.18.35.235:28614 SIP/2.0 Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport From: pepsi coke sip:12345678...@79.80.x.x:5678;tag=as42ec768c To: sip:a...@116.18.35.235:28614 Contact: sip:12345678...@79.80.x.x:5678 Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jul 2010 15:10:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 9626 9626 IN IP4 79.80.x.x s=session c=IN IP4 79.80.x.x t=0 0 m=audio 16238 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jul 21 11:10:22] WARNING[23814]: chan_sip.c:2872 sip_call: Setting auto-congest time to 15000 ms. -- Called a...@116.18.35.235:28614 ast-server*CLI --- SIP read from 116.18.35.235:28614 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678 Contact: sip:a...@116.18.35.235:28614 To: sip:a...@116.18.35.235:28614;tag=d54e632c From: pepsi cokesip:12345678...@79.80.x.x:5678;tag=as42ec768c Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x CSeq: 102 INVITE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 - --- (9 headers 0 lines) --- -- SIP/116.18.35.235:28614-007f4660 is ringing ast-server*CLI --- SIP read from 116.18.35.235:28614 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678 Contact: sip:a...@116.18.35.235:28614 To: sip:a...@116.18.35.235:28614;tag=d54e632c From: pepsi cokesip:12345678...@79.80.x.x:5678;tag=as42ec768c Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 185 v=0 o=- 6 2 IN IP4 192.168.0.12 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.0.12 t=0 0 m=audio 55246 RTP/AVP 8 0 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (11 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP
Re: [asterisk-users] IAX authentication oddity - Known issue? Fixed?
On 28 July 2010 17:32, Tilghman Lesher tles...@digium.com wrote: On Wednesday 28 July 2010 06:49:01 Steve Davies wrote: [snip] to avoid repetition below I don't see a 'type' argument to either of the above, so neither of these would at all be used. That said, you're assuming that the deny and allow determine who is allowed to be user2. That's incorrect. They permit what packets will even reach user2, and a registration needs to occur for the host address to become something other than 0.0.0.0 (which is the default, unless you have a defaultip parameter). Hence, user2 won't match anything at all until a registration packet comes in that passes your deny/allow ACL. Sorry, I missed the type=friend off both examples. Too busy cleaning up the example :( Perhaps it is better to describe what I want to achieve first... We want an primarily inbound IAX config that allows un-authenticated connections from a specified range of IP addresses. The remote party is required to use a username. I understood from the VoIP WiKi that if a username is provided by the caller in the NEW packet, and the permit/deny list allows the packet, that the following would be okay: [user2] type=user --- type=friend should be ok too ? username=user2 secret= context=context2 host=dynamic --- don't care placeholder. Is this bad? deny=0.0.0.0/0.0.0.0 allow=10.2.3.0/255.255.255.0 A channel is created called IAX/10.2.3.1-xxx in this example. What I am pretty sure I observed is that if I ALSO have the following configured: [user1] type=user username=user1 secret=a-secret context=context1 host=10.2.3.1 --- Specifically an address from the permit range in [user2] When a call arrives from IP address 10.2.3.1 with a username of user2, then [user2] is used for authentication, but the call proceeds using [user1] and a channel name of IAX/user1-xxx after authentication is complete. In the example above this meant password-free access to a password protected context. I appreciate this is an odd claim, and I will try to reproduce it using both 1.2 and 1.6 asterisk builds, in the meantime I was wondering if this was a known issue - Sounds like it isn't. PS. I think there is a 99% chance that I mis-interpreted the results, and this is not a real problem, but that 1% chance is why I am sending the email :) Thanks, Steve -- _ -- 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] IAX authentication oddity - Known issue? Fixed?
On 07/28/2010 11:32 AM, Tilghman Lesher wrote: They permit what packets will even reach user2 It should also be pointed out that the config option is permit, and not allow. -- _ -- 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] fail2ban and pf
On Wed, Jul 28, 2010 at 9:03 AM, Kyle Kienapfel doctor.w...@gmail.com wrote: On Wed, Jul 28, 2010 at 6:38 AM, Randy R randulo2...@gmail.com wrote: Hi, Since f2b is one of the topics du jour here, I was wondering if someone would mind telling me what these pf stats mean: Evaluations: 964303 Packets: 12176 Bytes: 648408 States: 0 80 or so blocked hosts? 964303/12176=~79.19 evaluations being more than packets looks like its going through multiple rules to think about packets. White listing your itsp and other traffic you know you like; before the list of banned computers might reduce that a bit. 80-100 certainly sound about right. While the essentials (MTA, important users, me...) are listed you're right, I should whitelist blocks and frequent accessors I know to be ok. There is a known good list already. Thanks! /r -- _ -- 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] 3G-324M Open Source
Hi We need to evaluate some open source project that supports 3G-324M on top of Asterisk. What do you recommend ? What has been your experience ? Thanks. regards, Anita Hall, Simmortel. -- _ -- 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] IAX authentication oddity - Known issue? Fixed?
On Wednesday 28 July 2010 12:18:04 Steve Davies wrote: When a call arrives from IP address 10.2.3.1 with a username of user2, then [user2] is used for authentication, but the call proceeds using [user1] and a channel name of IAX/user1-xxx after authentication is complete. In the example above this meant password-free access to a password protected context. I appreciate this is an odd claim, and I will try to reproduce it using both 1.2 and 1.6 asterisk builds, in the meantime I was wondering if this was a known issue - Sounds like it isn't. It definitely sounds like a bug, but I'd be interested to hear if this is still a problem in both 1.4 and 1.6.2. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Subscribe Problem - Zombie Channel
dotnetdub schrieb: Hi List, snip core show channels Channel Location State Application(Data) SIP/102--08e1 *...@from-inside Down(None) SIP/102--08d6 *...@from-inside Ring(None) SIP/102--08d7 *...@from-inside Ring(None) 3 active channels 0 active calls The only way to free them up is to force a restart. restart now Any clues on how I can debug this and try to sort it or even if anyone has come across this. Many thanks in advance. Brian hello, you should recompile asterisk with DEBUG CHANNEL LOCKS flag and i think you will see some locks when this happens with core show locks. how do you make the pickup? do you use an extension *8 for this, or just the feature for pickup in features conf? best regards steve -- _ -- 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 unresponsive
We are running asteriskNow 1.4.18 and after a few days it becomes unresponsive and inbound INVITEs timeout. We just reboot the box to resolve it. But it seems to be occurring more regularly now. I am hesitant to move to latest version, but will do if needed. Any guidance or troubleshooting modes I may use will be helpful. -- _ -- 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] Asterisk unresponsive
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo Subject: [asterisk-users] Asterisk unresponsive We are running asteriskNow 1.4.18 and after a few days it becomes unresponsive and inbound INVITEs timeout. We just reboot the box to resolve it. But it seems to be occurring more regularly now. Assuming you aren't around the clock operation, why not just set up a cron to do an asterisk -rx restart now at say 4.00 am local time each day Crontab -e 4 0 * * * /usr/sbin/asterisk -rx restart now -- _ -- 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] Asterisk unresponsive
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo Subject: [asterisk-users] Asterisk unresponsive We are running asteriskNow 1.4.18 and after a few days it becomes unresponsive and inbound INVITEs timeout. We just reboot the box to resolve it. But it seems to be occurring more regularly now. Assuming you aren't around the clock operation, why not just set up a cron to do an asterisk -rx restart now at say 4.00 am local time each day Crontab -e Correction 0 4 * * * /usr/sbin/asterisk -rx restart now -- _ -- 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] Why do Zaptel calls drop all of a sudden? Could busy detect be the problem?
Hi Guys, I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: [Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy, avgtone: 75, avgsilence 135 [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing [...@macro-dialout-trunk:1] Macro(SIP/2111-b6a400b0, hangupcall|) in new stack [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing [...@macro-hangupcall:1] GotoIf(SIP/2111-b6a400b0, 1?skiprg) in new stack [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Goto (macro-hangupcall,s,4) This is running 1.4.26.1 (Elastix) Should I turn of busy detect in chan_dahdi.conf? or is this a known bug and has a workaround? Thanks, Bruce -- _ -- 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] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Subject: [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem? I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: Could be callwaiting? -- _ -- 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] Asterisk unresponsive
From: Ujjval Karihaloo We are running asteriskNow 1.4.18 and after a few days it becomes unresponsive and inbound INVITEs timeout. We just reboot the box to resolve it. But it seems to be occurring more regularly now. On Wed, 28 Jul 2010, Danny Nicholas wrote: Assuming you aren’t “around the clock” operation, why not just set up a cron to do an asterisk –rx “restart now” at say 4.00 am local time each day Crontab –e Correction 0 4 * * * /usr/sbin/asterisk –rx “restart now” Not a big fan of band-aids, but I understand they have their place. Will restarting Asterisk make it responsive or do you have an OS issue? A very small nit -- single quotes should be infinitesimally more efficient because the shell will not try to do parameter substitution within the quoted string. If the intent is to schedule a job to be run daily, how about @daily instead of 0 4 * * * -- 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
Re: [asterisk-users] Subscribe Problem - Zombie Channel
On 28 July 2010 21:42, Stefan Schmidt s...@sil.at wrote: dotnetdub schrieb: Hi List, snip core show channels Channel Location State Application(Data) SIP/102--08e1 *...@from-inside Down(None) SIP/102--08d6 *...@from-inside Ring(None) SIP/102--08d7 *...@from-inside Ring(None) 3 active channels 0 active calls The only way to free them up is to force a restart. restart now Any clues on how I can debug this and try to sort it or even if anyone has come across this. Many thanks in advance. Brian hello, you should recompile asterisk with DEBUG CHANNEL LOCKS flag and i think you will see some locks when this happens with core show locks. how do you make the pickup? do you use an extension *8 for this, or just the feature for pickup in features conf? best regards steve Hi Steve, Thanks for the reply. We have: pickupexten = *8; Configure the pickup extension. Default is *8 in features.conf. I will recompile on one of the sites this happens on. It's really odd, can go for weeks without this happening and then a customer will report to me that their extension is showing in use and I will login and there can be two or three of these locks. On one site it actually makes asterisk impossible to stop and I need to kill -9 We have stuck with version 1.4.22 as it has been so solid for us, no dumps or deadlocks etc. We have tried to move to 1.4.25 and 1.4.29 but would experience random weirdness that we just don't get with this version. When recompiled with this flag and if indeed it does show locks, what would be the next step? Thanks for your help. Brian -- _ -- 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] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?
Hmmwhat about call waiting? You mean, when a call comes in on that specific line, it generate two beep tones and hence the system hangs up thinking it's end of the call? Interesting!!! If it is call-waiting do I have to set all of the following off for it to not give me problem again: *callwaiting=yes* *usecallingpres=yes* *callwaitingcallerid=yes* *threewaycalling=yes* *transfer=yes* *canpark=yes* *cancallforward=yes* *busydetect=yes* *busycount=3* Please elaborate a bit if I am off-topic. Regards, Bruce On Wed, Jul 28, 2010 at 5:38 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Subject:* [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem? I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: Could be callwaiting? -- _ -- 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] Asterisk unresponsive
We are running asteriskNow 1.4.18 and after a few days it becomes unresponsive and inbound INVITEs timeout. Search this list for DNS. Philipp -- _ -- 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] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem?
Furthermore, these are lines in Hunt, so, I am not sure if Call-Waiting is turned ON on these lines at all. But it's definitely an interesting idea. On Wed, Jul 28, 2010 at 5:54 PM, bruce bruce bruceb...@gmail.com wrote: Hmmwhat about call waiting? You mean, when a call comes in on that specific line, it generate two beep tones and hence the system hangs up thinking it's end of the call? Interesting!!! If it is call-waiting do I have to set all of the following off for it to not give me problem again: *callwaiting=yes* *usecallingpres=yes* *callwaitingcallerid=yes* *threewaycalling=yes* *transfer=yes* *canpark=yes* *cancallforward=yes* *busydetect=yes* *busycount=3* Please elaborate a bit if I am off-topic. Regards, Bruce On Wed, Jul 28, 2010 at 5:38 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Subject:* [asterisk-users] Why do Zaptel calls drop all of a sudden? Couldbusy detect be the problem? I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: Could be callwaiting? -- _ -- 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
[asterisk-users] app_swift.c:338 engine: Failed to set voice
Hello. I'm trying to set TTS with Cepstral and Swift but can't get it to work. I get this error when testing it: -- SIP/101- Playing 'welcome.gsm' (language 'es') -- Executing [...@local-calls:3] Swift(SIP/101-, Hello this is ceptral) in new stack [Jul 28 18:29:16] NOTICE[5191]: app_swift.c:304 engine: Text to Speak : Hello this is ceptral [Jul 28 18:29:16] ERROR[5191]: app_swift.c:338 engine: Failed to set voice. I'm using: asterisk*CLI core show version Asterisk 1.6.1.18 built by root @ optimum-asterisk on a i686 running Linux on 2010-04-10 01:42:25 UTC I googled around but, there isnt a real solution I could find. Any suggestions? Thanks in advanced for your help. -- _ -- 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] Asterisk unresponsive
Good call! I was just reading this thread and was preparing to write a reply mentioning DNS and SIP channel lockups... Basically, OP, Asterisk's SIP channels don't like not being able to do timely DNS queries, so you end up with a very unresponsive Asterisk server if you don't have local DNS caching [and][or] solid fast DNS servers available to you 24-7 I've been exactly where you are now, trust me, at first those lockups were causing me to lose sleep and think I was NUTS! Here's a helpful search: http://www.google.com/search?ie=UTF-8oe=UTF-8sourceid=navclientgfns=1q=asterisk-users+dns+sip+lockup Cheers, Sherwood McGowan ...I've been working with VoIP for almost 10 years now!?!?!AUUGH! On Wed, Jul 28, 2010 at 4:54 PM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: We are running asteriskNow 1.4.18 and after a few days it becomes unresponsive and inbound INVITEs timeout. Search this list for DNS. Philipp -- _ -- 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] Asterisk Crashes - Segmentation Fault
Manmohan wrote: I can see the path does exists but i cant see any recordings happening inn there. There are no files in it Following is the output: /var/lib/asterisk/sounds drwxrwxrwx 2 asterisk apache 4096 Jun 27 20:54 conf-recordings I hope m understandly this correctly but m sure m missing something here ;-) You did understand, and we have eliminated another of the possible issues. Are you assigning an admin pin to these conferences? There is a patch that allows recording pinless concenferences, but is has oddly not been merged yet. Try setting an admin pin. If that does not work, send the CLI output with core set verbose 5 as you dial in to the conference. Dan -- _ -- 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] Random DTMF Tones Only on heard on ATA
SIP/5211 is a Grandstream device. Did not add relaxdtmf=no, but sip show settings verifies it's already set to no. Fat fingered the version, it should have said 1.6.2.6 through 1.6.2.10 Travis On Wed, Jul 28, 2010 at 3:12 AM, Benny Amorsen benny+use...@amorsen.dkbenny%2buse...@amorsen.dk wrote: Travis Langhals tra...@netitek.com writes: [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin '1' received on SIP/5211-0078 Is SIP/5211 a Linksys or a Grandstream or something else? Do you have relaxdtmf=no? Also, your Asterisk version numbers are incorrect. Do you mean 1.6.2.10? /Benny -- _ -- 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] Random DTMF Tones Only on heard on ATA
Sorry, I came into this late...what codec is the device using, and is the audio being trascoded? Back at Voxitas, we had a couple of customers complain about random DTMF tones coming across their line, and Asterisk WAS actually hearing DTMF tones...want to know what it was?. In that particular case (just a place to start looking) it was G729 on customer ATAs (don't remember the models)Here's the freaky thingIt only happened with CERTAIN people talking on the phone...IIRC, we determined that the ATA's G729 processor was mistaking certain audio frequencies in the speaker's voice and believing it was a DTMF tone from the analog device and sending the appropriate DTMF signal to our servers... I'm sorry, I don't remember how we fixed it...I think we did some audio tweaking (advanced ATA config, input level, out level, etc..), be we may have just ended up having to tell that client to not use G729 on those ATAs This _MAY_ happen with other codecs, but I think it's mainly either G729..maybe primarily transcoding? NERDY FUn Crap below: capture SIP and RTP between your Asterisk and an offending device (writing to a file)then start doing everything you can to cause the DTMF issue to occur. NOW, open your capture in wireshark...dump the RTP payload to a file and open that file in an audio editor Now, go through the wireshark capture...see if you see any DTMF events (if rfc2833 it'll be an RTP EVENT, if SIP INFO, it'll be a sip info, and if you're using inband **SHUDDER** you can just listen to the audio).note the time in seconds from the beginning of the audio stream whenever a DTMF event occurs, and then go to that spot in the audio fileIf you're feeling REALLY frisky, do a frequency analysis...I'll bet you'll see that the voice that is speaking at the time of the DTMF event on your various captures will have a frequency range in common...a very close range...maybe look up DTMF tone definition and get the freqs(did itmore detail than even I feel like doing right now :D) Cheers, Sherwood McGowan On Wed, Jul 28, 2010 at 6:43 PM, Travis Langhals tra...@netitek.com wrote: SIP/5211 is a Grandstream device. Did not add relaxdtmf=no, but sip show settings verifies it's already set to no. Fat fingered the version, it should have said 1.6.2.6 through 1.6.2.10 Travis On Wed, Jul 28, 2010 at 3:12 AM, Benny Amorsen benny+use...@amorsen.dk wrote: Travis Langhals tra...@netitek.com writes: [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin '1' received on SIP/5211-0078 Is SIP/5211 a Linksys or a Grandstream or something else? Do you have relaxdtmf=no? Also, your Asterisk version numbers are incorrect. Do you mean 1.6.2.10? /Benny -- _ -- 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] app_swift.c:338 engine: Failed to set voice
On 7/28/2010 6:22 PM, Landy Landy wrote: [Jul 28 18:29:16] NOTICE[5191]: app_swift.c:304 engine: Text to Speak : Hello this is ceptral [Jul 28 18:29:16] ERROR[5191]: app_swift.c:338 engine: Failed to set voice. Do you have cepstral installed and have the voice(s) registered ? try: swift --voices assuming swift is installed an a valid voice is registered, what happens when you type: swift Test Message -o /tmp/file.wav is /tmp/file.wav created ? does it play ? what is the output of: grep ^[a-z] /etc/asterisk/swift.conf somewhere should say voice=X. Is that voice installed as per the above swift --voices command ? also, if you're going to be dialing digits with swift, you'll probably run into detection issues unless you use my patch at http://jeremy.kister.net/code/app_swift-1.6.2.patch -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] app_swift.c:338 engine: Failed to set voice
Do you have cepstral installed and have the voice(s) registered ? try: swift --voices asterisk:~# swift --voices Swift command-line synthesis program Version 5.1.0 of July 2008 Copyright (c) 2000-2006, Cepstral LLC. Voice | Version | Lic? | Gender | Age | Language | Sample Rate ---|-|--||-|--| Marta | 5.1.0 | No | female | 30 | Americas Spanish | 16000 Hz assuming swift is installed an a valid voice is registered, what happens when you type: swift Test Message -o /tmp/file.wav is /tmp/file.wav created ? does it play ? This creates the file and if I download it to my machine I can listen to it. what is the output of: grep ^[a-z] /etc/asterisk/swift.conf asterisk:/home/landysaccount# grep ^[a-z] /etc/asterisk/swift.conf buffer_size=65535 goto_exten=no voice=Marta-8kHz|David-8kHz somewhere should say voice=X. Is that voice installed as per the above swift --voices command ? also, if you're going to be dialing digits with swift, you'll probably run into detection issues unless you use my patch at http://jeremy.kister.net/code/app_swift-1.6.2.patch I had to patch that file in order for me to be able to install swift. -- _ -- 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 stopped after Internet connection dropped ?! Asterisk 1.4.26.1
Hi Everyone, This is probably more related to Linux than to Asterisk. Analogue channels on a system were un-responsive on Monday morning. Apparently something happened over the weekend and the router went off or it lost it's DSL connection. [Jul 23 22:50:01] VERBOSE[12437] logger.c: -- Remote UNIX connection [Jul 23 22:50:01] VERBOSE[27087] logger.c: -- Remote UNIX connection disconnected [Jul 23 22:55:01] VERBOSE[12437] logger.c: -- Remote UNIX connection [Jul 23 22:55:01] VERBOSE[27093] logger.c: -- Remote UNIX connection disconnected [Jul 23 23:00:01] VERBOSE[12437] logger.c: -- Remote UNIX connection [Jul 23 23:00:02] VERBOSE[27099] logger.c: -- Remote UNIX connection disconnected [Jul 26 09:22:59] VERBOSE[3529] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log [Jul 26 09:22:59] VERBOSE[3529] logger.c: Asterisk Dynamic Loader Starting: [Jul 26 09:22:59] VERBOSE[3529] logger.c: == Parsing '/etc/asterisk/modules.conf': [Jul 26 09:22:59] VERBOSE[3529] logger.c: Found [Jul 26 09:22:59] VERBOSE[3529] logger.c: == Parsing '/etc/asterisk/dnsmgr.conf': [Jul 26 09:22:59] VERBOSE[3529] logger.c: Found [Jul 26 09:22:59] VERBOSE[3529] logger.c: == Parsing '/etc/asterisk/http.conf': [Jul 26 09:22:59] VERBOSE[3529] logger.c: Found See the jump from Jul 23rd to Jul 26th. Is this an indication of Asterisk being down? I don't see any of that but yet no calls are on the report for July 24th and 25th indicating to me that Analogue channels, or Asterisk, or the server was down during this time as this office always receives calls on the weekend to the IVR. Where are the logs for eth0 so that I can check to see why this happened and if indeed it was a drop in internet connection. If so, and this is the known bug for Asterisk stop working due to internet drop, why is it not listed in the log file posted above? Thanks, -- _ -- 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] app_swift.c:338 engine: Failed to set voice
On 7/28/2010 8:33 PM, Landy Landy wrote: asterisk:/home/landysaccount# grep ^[a-z] /etc/asterisk/swift.conf buffer_size=65535 goto_exten=no voice=Marta-8kHz|David-8kHz afaik, the voice parameter is simply the default voice when not specified via the swift binary or the Swift asterisk command. even if it's not, you don't have David registered. try making that: voice=Marta (or possibly: voice=Marta-8kHz) then restart asterisk and give it another shot. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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] app_swift.c:338 engine: Failed to set voice
Jeremy, Thanks a lot that helped and solved the problem. I had it as: voice=Marta-8kHz before and that didn't work and now changed it to voice=Marta. Thanks. I apreciate it. --- On Wed, 7/28/10, Jeremy Kister asterisk...@jeremykister.com wrote: From: Jeremy Kister asterisk...@jeremykister.com Subject: Re: [asterisk-users] app_swift.c:338 engine: Failed to set voice To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, July 28, 2010, 9:08 PM On 7/28/2010 8:33 PM, Landy Landy wrote: asterisk:/home/landysaccount# grep ^[a-z] /etc/asterisk/swift.conf buffer_size=65535 goto_exten=no voice=Marta-8kHz|David-8kHz afaik, the voice parameter is simply the default voice when not specified via the swift binary or the Swift asterisk command. even if it's not, you don't have David registered. try making that: voice=Marta (or possibly: voice=Marta-8kHz) then restart asterisk and give it another shot. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- 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
[asterisk-users] spam blacklist
Just a note, the asterisk mailing list server continually gets blacklisted over at http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering mail to spamtraps. Perhaps something needs to be looked into... Regards, Sam -- _ -- 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] spam blacklist
My guess is on spammers signing up the spamtraps for mailing lists ;) On Wed, Jul 28, 2010 at 6:45 PM, Sam aster...@net153.net wrote: Just a note, the asterisk mailing list server continually gets blacklisted over at http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering mail to spamtraps. Perhaps something needs to be looked into... Regards, Sam -- _ -- 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] spam blacklist
SIP wrote: what can you do ? simple discard spam don't bounce it. On 7/28/10 9:45 PM, Sam wrote: Just a note, the asterisk mailing list server continually gets blacklisted over at http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering mail to spamtraps. Perhaps something needs to be looked into... Regards, Sam Spammers sign up to the Asterisk mailing list and send spam once in a while. My spam filter rejects it, and bounces the emails back to the Asterisk list, which then drops me from the list because it got a single bounce. Bit of a pain in the left ventricle, really, but what can you do. N. -- _ -- 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] spam blacklist
On 7/28/10 9:45 PM, Sam wrote: Just a note, the asterisk mailing list server continually gets blacklisted over at http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering mail to spamtraps. Perhaps something needs to be looked into... Regards, Sam Spammers sign up to the Asterisk mailing list and send spam once in a while. My spam filter rejects it, and bounces the emails back to the Asterisk list, which then drops me from the list because it got a single bounce. Bit of a pain in the left ventricle, really, but what can you do. N. -- _ -- 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] Asterisk stopped after Internet connection dropped ?! Asterisk 1.4.26.1
On Wed, Jul 28, 2010 at 6:06 PM, bruce bruce bruceb...@gmail.com wrote: Hi Everyone, This is probably more related to Linux than to Asterisk. Analogue channels on a system were un-responsive on Monday morning. Apparently something happened over the weekend and the router went off or it lost it's DSL connection. [Jul 23 22:50:01] VERBOSE[12437] logger.c: -- Remote UNIX connection [Jul 23 22:50:01] VERBOSE[27087] logger.c: -- Remote UNIX connection disconnected [Jul 23 22:55:01] VERBOSE[12437] logger.c: -- Remote UNIX connection [Jul 23 22:55:01] VERBOSE[27093] logger.c: -- Remote UNIX connection disconnected [Jul 23 23:00:01] VERBOSE[12437] logger.c: -- Remote UNIX connection [Jul 23 23:00:02] VERBOSE[27099] logger.c: -- Remote UNIX connection disconnected [Jul 26 09:22:59] VERBOSE[3529] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log [Jul 26 09:22:59] VERBOSE[3529] logger.c: Asterisk Dynamic Loader Starting: [Jul 26 09:22:59] VERBOSE[3529] logger.c: == Parsing '/etc/asterisk/modules.conf': [Jul 26 09:22:59] VERBOSE[3529] logger.c: Found [Jul 26 09:22:59] VERBOSE[3529] logger.c: == Parsing '/etc/asterisk/dnsmgr.conf': [Jul 26 09:22:59] VERBOSE[3529] logger.c: Found [Jul 26 09:22:59] VERBOSE[3529] logger.c: == Parsing '/etc/asterisk/http.conf': [Jul 26 09:22:59] VERBOSE[3529] logger.c: Found See the jump from Jul 23rd to Jul 26th. Is this an indication of Asterisk being down? I don't see any of that but yet no calls are on the report for July 24th and 25th indicating to me that Analogue channels, or Asterisk, or the server was down during this time as this office always receives calls on the weekend to the IVR. Where are the logs for eth0 so that I can check to see why this happened and if indeed it was a drop in internet connection. If so, and this is the known bug for Asterisk stop working due to internet drop, why is it not listed in the log file posted above? Thanks, -- _ -- 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 What sort of logging output do you normally get when asterisk starts? without more information I'd guess that the server was sleeping... -- _ -- 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] Asterisk Crashes - Segmentation Fault
Hi Dan, Following is the output for core set verbose 5, also i am really not able to get on the admin pin thing? Do you mean, that with admin pin configured we cant use recording? LinuxTest*CLI core set verbose 5 Verbosity was 3 and is now 5 == Using SIP RTP CoS mark 5 -- Executing [...@callman_incoming:1] MeetMe(SIP/callman02-0002, ) in new stack -- SIP/callman02-0002 Playing 'conf-getconfno.ulaw' (language 'en') == Parsing '/etc/asterisk/meetme.conf': == Found -- Created MeetMe conference 1023 for conference '77972' -- SIP/callman02-0002 Playing 'conf-getpin.ulaw' (language 'en') Starting recording of MeetMe Conference 77972 into file .. -- SIP/callman02-0002 Playing 'vm-rec-name.ulaw' (language 'en') [Jul 29 09:16:19] WARNING[25049]: file.c:1160 ast_writefile: No such format '' -- SIP/callman02-0002 Playing 'beep.ulaw' (language 'en') -- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-77972-1 format: sln, 0x9bea628 -- User ended message by pressing # -- SIP/callman02-0002 Playing 'auth-thankyou.ulaw' (language 'en') -- SIP/callman02-0002 Playing 'conf-onlyperson.ulaw' (language 'en') == Using SIP RTP CoS mark 5 -- Executing [...@callman_incoming:1] MeetMe(SIP/callman02-0003, ) in new stack -- SIP/callman02-0003 Playing 'conf-getconfno.ulaw' (language 'en') -- SIP/callman02-0003 Playing 'conf-getpin.ulaw' (language 'en') Starting recording of MeetMe Conference 77972 into file .. -- SIP/callman02-0003 Playing 'vm-rec-name.ulaw' (language 'en') -- SIP/callman02-0003 Playing 'beep.ulaw' (language 'en') -- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-77972-2 format: sln, 0x9bea628 -- User ended message by pressing # -- SIP/callman02-0003 Playing 'auth-thankyou.ulaw' (language 'en') -- DAHDI/pseudo-736798397 Playing '/var/spool/asterisk/meetme/meetme-username-77972-2.slin' (language 'en') -- DAHDI/pseudo-736798397 Playing 'conf-hasjoin.ulaw' (language 'en') -- SIP/callman02-0003 Playing 'conf-placeintoconf.ulaw' (language 'en') == Spawn extension (callman_incoming, 493, 1) exited non-zero on 'SIP/callman02-0002' -- Executing [...@callman_incoming:1] Set(SIP/callman02-0002, CDR(bookId)=) in new stack -- Executing [...@callman_incoming:2] Set(SIP/callman02-0002, CDR(CIDnum)=281) in new stack -- Executing [...@callman_incoming:3] Set(SIP/callman02-0002, CDR(CIDname)=Manmohan Singh Jandu) in new stack -- DAHDI/pseudo-736798397 Playing '/var/spool/asterisk/meetme/meetme-username-77972-1.slin' (language 'en') -- SIP/callman02-0003 Playing 'conf-leaderhasleft.ulaw' (language 'en') -- DAHDI/pseudo-736798397 Playing 'conf-hasleft.ulaw' (language 'en') -- Hungup 'DAHDI/pseudo-923268627' -- Hungup 'DAHDI/pseudo-736798397' == Spawn extension (callman_incoming, 493, 1) exited non-zero on 'SIP/callman02-0003' -- Executing [...@callman_incoming:1] Set(SIP/callman02-0003, CDR(bookId)=) in new stack -- Executing [...@callman_incoming:2] Set(SIP/callman02-0003, CDR(CIDnum)=115) in new stack -- Executing [...@callman_incoming:3] Set(SIP/callman02-0003, CDR(CIDname)=cipc) in new stack On Thu, Jul 29, 2010 at 2:39 AM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: I can see the path does exists but i cant see any recordings happening inn there. There are no files in it Following is the output: /var/lib/asterisk/sounds drwxrwxrwx 2 asterisk apache 4096 Jun 27 20:54 conf-recordings I hope m understandly this correctly but m sure m missing something here ;-) You did understand, and we have eliminated another of the possible issues. Are you assigning an admin pin to these conferences? There is a patch that allows recording pinless concenferences, but is has oddly not been merged yet. Try setting an admin pin. If that does not work, send the CLI output with core set verbose 5 as you dial in to the conference. Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
Also following is what i am putting in lib/define.php define (RECORDING_PATH, /var/lib/asterisk/sounds/conf-recordings/); On Thu, Jul 29, 2010 at 9:20 AM, Manmohan Singh Jandu manmoha...@gmail.comwrote: Hi Dan, Following is the output for core set verbose 5, also i am really not able to get on the admin pin thing? Do you mean, that with admin pin configured we cant use recording? LinuxTest*CLI core set verbose 5 Verbosity was 3 and is now 5 == Using SIP RTP CoS mark 5 -- Executing [...@callman_incoming:1] MeetMe(SIP/callman02-0002, ) in new stack -- SIP/callman02-0002 Playing 'conf-getconfno.ulaw' (language 'en') == Parsing '/etc/asterisk/meetme.conf': == Found -- Created MeetMe conference 1023 for conference '77972' -- SIP/callman02-0002 Playing 'conf-getpin.ulaw' (language 'en') Starting recording of MeetMe Conference 77972 into file .. -- SIP/callman02-0002 Playing 'vm-rec-name.ulaw' (language 'en') [Jul 29 09:16:19] WARNING[25049]: file.c:1160 ast_writefile: No such format '' -- SIP/callman02-0002 Playing 'beep.ulaw' (language 'en') -- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-77972-1 format: sln, 0x9bea628 -- User ended message by pressing # -- SIP/callman02-0002 Playing 'auth-thankyou.ulaw' (language 'en') -- SIP/callman02-0002 Playing 'conf-onlyperson.ulaw' (language 'en') == Using SIP RTP CoS mark 5 -- Executing [...@callman_incoming:1] MeetMe(SIP/callman02-0003, ) in new stack -- SIP/callman02-0003 Playing 'conf-getconfno.ulaw' (language 'en') -- SIP/callman02-0003 Playing 'conf-getpin.ulaw' (language 'en') Starting recording of MeetMe Conference 77972 into file .. -- SIP/callman02-0003 Playing 'vm-rec-name.ulaw' (language 'en') -- SIP/callman02-0003 Playing 'beep.ulaw' (language 'en') -- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-77972-2 format: sln, 0x9bea628 -- User ended message by pressing # -- SIP/callman02-0003 Playing 'auth-thankyou.ulaw' (language 'en') -- DAHDI/pseudo-736798397 Playing '/var/spool/asterisk/meetme/meetme-username-77972-2.slin' (language 'en') -- DAHDI/pseudo-736798397 Playing 'conf-hasjoin.ulaw' (language 'en') -- SIP/callman02-0003 Playing 'conf-placeintoconf.ulaw' (language 'en') == Spawn extension (callman_incoming, 493, 1) exited non-zero on 'SIP/callman02-0002' -- Executing [...@callman_incoming:1] Set(SIP/callman02-0002, CDR(bookId)=) in new stack -- Executing [...@callman_incoming:2] Set(SIP/callman02-0002, CDR(CIDnum)=281) in new stack -- Executing [...@callman_incoming:3] Set(SIP/callman02-0002, CDR(CIDname)=Manmohan Singh Jandu) in new stack -- DAHDI/pseudo-736798397 Playing '/var/spool/asterisk/meetme/meetme-username-77972-1.slin' (language 'en') -- SIP/callman02-0003 Playing 'conf-leaderhasleft.ulaw' (language 'en') -- DAHDI/pseudo-736798397 Playing 'conf-hasleft.ulaw' (language 'en') -- Hungup 'DAHDI/pseudo-923268627' -- Hungup 'DAHDI/pseudo-736798397' == Spawn extension (callman_incoming, 493, 1) exited non-zero on 'SIP/callman02-0003' -- Executing [...@callman_incoming:1] Set(SIP/callman02-0003, CDR(bookId)=) in new stack -- Executing [...@callman_incoming:2] Set(SIP/callman02-0003, CDR(CIDnum)=115) in new stack -- Executing [...@callman_incoming:3] Set(SIP/callman02-0003, CDR(CIDname)=cipc) in new stack On Thu, Jul 29, 2010 at 2:39 AM, Dan Austin dan_aus...@phoenix.comwrote: Manmohan wrote: I can see the path does exists but i cant see any recordings happening inn there. There are no files in it Following is the output: /var/lib/asterisk/sounds drwxrwxrwx 2 asterisk apache 4096 Jun 27 20:54 conf-recordings I hope m understandly this correctly but m sure m missing something here ;-) You did understand, and we have eliminated another of the possible issues. Are you assigning an admin pin to these conferences? There is a patch that allows recording pinless concenferences, but is has oddly not been merged yet. Try setting an admin pin. If that does not work, send the CLI output with core set verbose 5 as you dial in to the conference. Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- Thanks Regards Manmohan Singh Jandu -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
