Re: [asterisk-users] Two Asterisks behind NAT and need to link them using IAX trunk
Hi Bilal, How can I use SSH in that senario? Is there a link that can help to understand what I have to install and to configure? I don't think SSH is a recommended approach. You can't run an IAX2 trunk over SSH (IAX2 used UDP and SSH only supports TCP port forwarding. http://www.securityfocus.com/infocus/1816 documents TCP port forwarding over SSH. As above in this thread have a suggested, you'll need to implement OpenVPN (TCP tunnel) over SSH in order to establish an IAX2 trunk. It is much simpler to just use OpenVPN and forget about SSH altogether. The additional overhead in an IAX2 over OpenVPN over SSH, coupled with the use of TCP for the SSH and OpenVPN tunnels, will cause more problems with voice quality. The documentation on openvpn.net is excellent. Try http://openvpn.net/static.html for quick guide using static pre-shared keys. Installation of openvpn on your Linux distribution should be a simple as: Ubuntu/Debian: apt-get install openvpn Redhat based: http://dag.wieers.com/packages/openvpn will give you an RPM Gentoo: emerge openvpn Others: use tarball and compile, or find appropriate package Good luck and feel free to email back if you have troubles. Regards, Chris Bennett ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
At 11:21 PM 1/18/2008, you wrote: I would suspect that your hardware is the cause of your problems. Running a production PBX system on a discarded desktop system is a really bad idea. I would seriously consider an upgrade to your hardware. Except that it's been running 1.2 for 2 years with no problems. It has to handle all of about 20 to 30 calls/day so it's not really under much load and I see no reason why it should be a problem. If not for the TDM404 I would probably try to put it on a NSLU2. What would you recommend I run it on? Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
At 03:35 PM 1/18/2008, you wrote: I'm running 1.4 in production on the following two systems: Tyan GT20 AMD 939 dual core. openSuSE x86_64 10.1 Celeron 2.4ghz RHEL 4... cheap server from ThePlanet from what I recall they use cheap cheap cheap consumer grade stuff. Not a single crash not a single issue. Yes I know, most of the people running 1.4 do it without problem today, I've tried and failed. I'll try the 1.6 beta Sunday or next weekend and see if that will run. I promise I'm not averse to trying early code, I've been beta testing something or another almost continuously since I figured out how to break Brief on a Netware network in 86 or so. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
At 02:20 PM 1/18/2008, you wrote: On Fri, Jan 18, 2008 at 12:20:56PM -0800, Ira wrote: At 11:53 AM 1/18/2008, you wrote: Although for some of us, or at least me, no version of 1.4 has run for more than 72 hours before generating a kernel panic. I've tried about 6 versions, the early ones were good for about 10 minutes, the latest one lasted 3 days. Sadly I'm still stuck using the latest 1.2. Kernel panics can be caused by buggy kernel code and / or bad hardware. Buggy userspace should not (by definition) be able to cause them. If userspace can, it's a kernel bug. So can you be more specific about those panics? Do you have traces from them? If I had any idea how I might go about that and if anyone had seemed to care I'd have done anything asked. The only Linux box I've ever touched is this one and I know just enough to build Asterisk and keep it alive. MS-DOS I can do anything in; Windows, close to anything; Linux, I'm qualified to turn it on and type yum update occasionally. I use MC for most everything. Not that I'm not interested, it's just not something I need for anything other than Asterisk. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Iax Encryption
Cavalera Claudio Luigi wrote: I guess what you are meaning here is it's easy to configure on asterisk side. Correct. So this encryption is now considered robust enough to be used in production? I certainly consider it that way. There should not be any problems with it. At our last developer's conference, we discussed some potential ways to improve it, but it should be fine as it is. I'm asking this because of comments I've found here: http://www.voip-info.org/wiki/index.php?page=IAX%20encryption about beta stage encryption. Content on the wiki is quite often incorrect and/or out of date, unfortunately. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nightly tarballs, would you use them?
Greetings, During the past week, there have been some requests for nightly tarballs to help making testing new Asterisk code easier. There was some debate as to whether they would be useful. The reason that they may not be useful is because you can get equivalent access to new code just by accessing the subversion repository directly. However, for one reason or another, some people would prefer to have a tarball. If this was available, would you be interested in it? If you just want to say yes or no for the sake of the poll, fell free to respond to me off-list. However, also fell free to respond here if you have more verbose comments on the topic that you would like to share. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nightly tarballs, would you use them?
Russell Bryant wrote: Greetings, During the past week, there have been some requests for nightly tarballs to help making testing new Asterisk code easier. There was some debate as to whether they would be useful. The reason that they may not be useful is because you can get equivalent access to new code just by accessing the subversion repository directly. However, for one reason or another, some people would prefer to have a tarball. If this was available, would you be interested in it? On occasion, yes. I think nightly tarballs could be quite useful. Whilst it's easy to check out from subversion directly, a nightly tarball provides a specific point of reference which can be helpful when trying to identify a problem. If we had a specific problem we were trying to fix, I would very likely grab the latest tarball and try it out. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call-out campaign variable problem
Hello!
Looks like list does not get this message. If it is duplicate sorry.
My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on
Fedora Core 4. I am making automatic call-out campaign with this setup
on 4 PRI. The scripts for this:
caller php script write this to outgoung folder:
fwrite($outfile,Channel: Zap/g1/$phonenumber\n);
fwrite($outfile,MaxRetries: 0\n);
fwrite($outfile,RetryTime: 5\n);
fwrite($outfile,WaitTime: 20\n);
fwrite($outfile,Context: 0100q\n);
fwrite($outfile,Callerid: $dbid\n);
fwrite($outfile,Extension: $phonenumber\n);
fwrite($outfile,Set: par_telszam=$phonenumber\n);
extensions.conf:
[0100q]
exten = _.,1,Wait(1)
exten = _.,n,Set(__TRIES=1)
exten = _.,n,Set(__FMT_DATE=%Y-%m-%d %H:%M:%S)
exten = _.,n,Set(__SZAM=${par_telszam})
exten = _.,n,System(echo -e
${SZAM}\,felvette\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_0.txt)
exten = _.,n,Playback(0100q_0)
exten = _.,n,System(echo -e
${SZAM}\,99\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_1v.txt)
exten = _.,n(valasztas),Set(TIMEOUT(response)=5)
exten = _.,n,Set(TIMEOUT(digit)=1)
exten = _.,n,Background(0100q_1)
exten = t,1,System(echo -e
${SZAM}\,timeout\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_1.txt)
exten = t,n,Goto(0100q_2,999,1)
exten = i,1,System(echo -e
${SZAM}\,invalid\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_1.txt)
exten = i,n,Goto(0100q_2,999,1)
exten = 1,1,System(echo -e
${SZAM}\,1\,${STRFTIME(${EPOCH},,${FMT_DATE})} /tmp/0100q_1.txt)
exten = 1,n,Goto(0100q_2,999,1)
exten = 2,1,System(echo -e
${SZAM}\,2\,${STRFTIME(${EPOCH},,${FMT_DATE})} /tmp/0100q_1.txt)
exten = 2,n,Goto(0100q_2,999,1)
exten = 3,1,System(echo -e
${SZAM}\,3\,${STRFTIME(${EPOCH},,${FMT_DATE})} /tmp/0100q_1.txt)
exten = 3,n,Goto(0100q_2,999,1)
exten = 9,1,System(echo -e
${SZAM}\,9\,${STRFTIME(${EPOCH},,${FMT_DATE})} /tmp/0100q_1v.txt)
exten = 9,n,GotoIf($[${TRIES} = 4.00]?0100q_2,999,1)
exten = 9,n,Set(__TRIES=${MATH(${TRIES}+1)})
exten = 9,n,Wait(1)
exten = 9,n,Goto(_.,valasztas)
[0100q_2]
exten = 999,1,Wait(1)
exten = 999,n,Background(0100q_2)
exten = t,1,System(echo -e
${SZAM}\,timeout\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_2.txt)
exten = t,n,Goto(0100q_9,999,1)
exten = i,1,System(echo -e
${SZAM}\,invalid\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_2.txt)
exten = i,n,Goto(0100q_9,999,1)
exten = 1,1,System(echo -e
${SZAM}\,1\,${STRFTIME(${EPOCH},,${FMT_DATE})} /tmp/0100q_2.txt)
exten = 1,n,Goto(0100q_9,999,1)
[0100q_9]
exten = 999,1,Wait(1)
exten = 999,n,System(echo -e
${SZAM}\,elkoszont\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_9.txt)
exten = 999,n,Playback(0100q_9)
exten = 999,n,Hangup
stats:
wc -l 0100q_0.txt = 14628
cut -d , -f 1 0100q_0.txt | sort | uniq -c -d | wc -l = 74
wc -l 0100q_1v.txt = 14300
cut -d , -f 1 0100q_1v.txt | sort | uniq -c -d | wc -l = 498
grep ,99, 0100q_1v.txt | cut -d , -f 1 | sort | uniq -c -d | wc -l = 66
cut -d , -f 1 0100q_1.txt | sort | uniq -c -d | wc -l = 0
same for 2 and 9
Txt format is number,string,date.
Caller script call every number once if call was successful. I
checked. Therefore there can not be duplicates in _0, there can not be
multiple 99 string for a number. Looks like there is some variable
problem but I did not find where is it. Because there is thousands of
successful calls the script should be correct I think.
Any idea why is it happen? Is it a bug or I am just blind?
bye,
a
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Re: [asterisk-users] Nightly tarballs, would you use them?
Per Jessen wrote: Russell Bryant wrote: Greetings, During the past week, there have been some requests for nightly tarballs to help making testing new Asterisk code easier. There was some debate as to whether they would be useful. The reason that they may not be useful is because you can get equivalent access to new code just by accessing the subversion repository directly. However, for one reason or another, some people would prefer to have a tarball. If this was available, would you be interested in it? On occasion, yes. I think nightly tarballs could be quite useful. Whilst it's easy to check out from subversion directly, a nightly tarball provides a specific point of reference which can be helpful when trying to identify a problem. If we had a specific problem we were trying to fix, I would very likely grab the latest tarball and try it out. /Per Jessen, Zürich In subversion can you specify what revision you want to check out so it is equally easy to know what version you want to test. I can agree that a nightly tarball is a bit more spoon feeding for none developer people. And to create a nightly tarball is a script and a cron jobb so the resources to maintain it should be low. And for the poll, I would unlikely use the tarball. /Mats ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nightly tarballs, would you use them?
On Sat, Jan 19, 2008 at 11:23:44AM +0100, Per Jessen wrote:
Russell Bryant wrote:
Greetings,
During the past week, there have been some requests for nightly
tarballs to help making testing new Asterisk code easier. There was
some debate as to whether they would be useful. The reason that they
may not be useful is because you can get equivalent access to new
code just by accessing the subversion repository directly. However,
for one reason or another, some people would prefer to have a tarball.
If this was available, would you be interested in it?
On occasion, yes.
I think nightly tarballs could be quite useful. Whilst it's easy to
check out from subversion directly, a nightly tarball provides a
specific point of reference which can be helpful when trying to
identify a problem.
svn co -r1 http://svn.digium.com/svn/asterisk/trunk asterisk-r1000
svn co -r'{2008-01-18}' http://svn.digium.com/svn/asterisk/trunk
asterisk-20080118
(use 'svn update' with the same -r switch in an existing copy, of
course)
http://svnbook.red-bean.com/en/1.4/svn.tour.revs.specifiers.html#svn.tour.revs.dates
If we had a specific problem we were trying to
fix, I would very likely grab the latest tarball and try it out.
The latest nightly tarball is not the latest SVN. Some problems may have
been fixed since.
svn co http://svn.digium.com/svn/asterisk/trunk asterisk-latest
Also, if your timezone is of the US, the nightly tarball may come in the
middle of your work day. Less of an issue for Europeans. More of an
issue for Indians and farther east.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Automatic call-out problem
On Fri, Jan 18, 2008 at 01:14:51PM +0100, Artifex Maximus wrote:
Hello!
My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on
Fedora Core 4. I am making automatic call-out campaign with this setup
on 4 PRI. The scripts for this:
caller php script write this to outgoung folder:
fwrite($outfile,Channel: Zap/g1/$phonenumber\n);
fwrite($outfile,MaxRetries: 0\n);
fwrite($outfile,RetryTime: 5\n);
fwrite($outfile,WaitTime: 20\n);
fwrite($outfile,Context: 0100q\n);
fwrite($outfile,Callerid: $dbid\n);
fwrite($outfile,Extension: $phonenumber\n);
fwrite($outfile,Set: par_telszam=$phonenumber\n);
extensions.conf:
[0100q]
exten = _.,1,Wait(1)
exten = _.,n,Set(__TRIES=1)
exten = _.,n,Set(__FMT_DATE=%Y-%m-%d %H:%M:%S)
exten = _.,n,Set(__SZAM=${par_telszam})
exten = _.,n,System(echo -e
${SZAM}\,felvette\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_0.txt)
What is this for? Why would you invent your own personal logging and not
use Asterisk's one?
Use a unique-enough message and Verbose. What you use is quite
inefficient.
Anyway, why not show us a trace from the CLI?
set verbose 3
And see what happens when you drop a call file.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
What you run it on is very much a function of how reliable you want the system to be. The better the hardware, the more reliable it will be. If you're running in a business environment, then I wouldn't recommend anything less than server grade - even if it's low end server grade. The company I work for supplies either Dell PowerEdge 860s (1RU servers that are similar in price to an upper mid-range desktop) or Dell PowerEdge 840s (tower cases that are similar in price to a mid-range desktop) Running on cheap hardware is a great way to cost yourself more in the long run - in lost productivity, lost sales and IT support. Ira wrote: At 11:21 PM 1/18/2008, you wrote: I would suspect that your hardware is the cause of your problems. Running a production PBX system on a discarded desktop system is a /really/ bad idea. I would seriously consider an upgrade to your hardware. Except that it's been running 1.2 for 2 years with no problems. It has to handle all of about 20 to 30 calls/day so it's not really under much load and I see no reason why it should be a problem. If not for the TDM404 I would probably try to put it on a NSLU2. What would you recommend I run it on? Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Automatic call-out problem
Hello!
Thanks for your answer!
On Jan 19, 2008 12:49 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Jan 18, 2008 at 01:14:51PM +0100, Artifex Maximus wrote:
My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on
Fedora Core 4. I am making automatic call-out campaign with this setup
on 4 PRI. The scripts for this:
caller php script write this to outgoung folder:
fwrite($outfile,Channel: Zap/g1/$phonenumber\n);
fwrite($outfile,MaxRetries: 0\n);
fwrite($outfile,RetryTime: 5\n);
fwrite($outfile,WaitTime: 20\n);
fwrite($outfile,Context: 0100q\n);
fwrite($outfile,Callerid: $dbid\n);
fwrite($outfile,Extension: $phonenumber\n);
fwrite($outfile,Set: par_telszam=$phonenumber\n);
extensions.conf:
[0100q]
exten = _.,1,Wait(1)
exten = _.,n,Set(__TRIES=1)
exten = _.,n,Set(__FMT_DATE=%Y-%m-%d %H:%M:%S)
exten = _.,n,Set(__SZAM=${par_telszam})
exten = _.,n,System(echo -e
${SZAM}\,felvette\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_0.txt)
What is this for? Why would you invent your own personal logging and not
use Asterisk's one?
We must know call is handled or not and which option callee choose for
a given question. It is an old script (back to * 1.0 era) and just
works till now. I only rewrite PHP script with Set in call file and
extension not using ${EXTEN} but ${par_telszam}. So the problem must
be there.
Use a unique-enough message and Verbose. What you use is quite
inefficient.
You are right but there is problem with Set through call file and that
is the question now. Grepping huge log file is efficient? Would be
great having persistent database connection in Asterisk and using
that. We had use MYSQL() but that was slow and complicated and finally
we switch to this kind of logging. Might not the best but working.
Anyway, why not show us a trace from the CLI?
set verbose 3
And see what happens when you drop a call file.
Here it is, I am masking out phone numbers:
astibm1*CLI set verbose 3
Verbosity is at least 3
-- Attempting call on Zap/g1/06x for [EMAIL PROTECTED]:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- Executing Wait(Zap/1-1, 1) in new stack
-- Executing Set(Zap/1-1, __TRIES=1) in new stack
-- Executing Set(Zap/1-1, __FMT_DATE=%Y-%m-%d %H:%M:%S) in new stack
-- Executing Set(Zap/1-1, __SZAM=06x) in new stack
-- Executing System(Zap/1-1, echo -e
06x,felvette,2008-01-19 13:16:33
/root/export/0100q_0.txt) in new stack
-- Executing Playback(Zap/1-1, 0100q_0) in new stack
-- Playing '0100q_0' (language 'hu')
-- Executing System(Zap/1-1, echo -e 06x,99,2008-01-19
13:17:10 /root/export/0100q_1v.txt) in new stack
-- Executing Set(Zap/1-1, TIMEOUT(response)=8) in new stack
-- Response timeout set to 8
-- Executing Set(Zap/1-1, TIMEOUT(digit)=1) in new stack
-- Digit timeout set to 1
-- Executing BackGround(Zap/1-1, 0100q_1) in new stack
-- Playing '0100q_1' (language 'hu')
== CDR updated on Zap/1-1
-- Executing System(Zap/1-1, echo -e 06x,1,2008-01-19
13:17:19 /root/export/0100q_1.txt) in new stack
-- Executing Goto(Zap/1-1, 0100q_2|999|1) in new stack
-- Goto (0100q_2,999,1)
-- Executing Wait(Zap/1-1, 1) in new stack
-- Executing BackGround(Zap/1-1, 0100q_2) in new stack
-- Playing '0100q_2' (language 'hu')
== CDR updated on Zap/1-1
-- Executing System(Zap/1-1, echo -e 06x,1,2008-01-19
13:17:25 /root/export/0100q_2.txt) in new stack
-- Executing Goto(Zap/1-1, 0100q_9|999|1) in new stack
-- Goto (0100q_9,999,1)
-- Executing Wait(Zap/1-1, 1) in new stack
-- Executing System(Zap/1-1, echo -e
06x,elkoszont,2008-01-19 13:17:26
/root/export/0100q_9.txt) in new stack
-- Executing Playback(Zap/1-1, 0100q_9) in new stack
-- Playing '0100q_9' (language 'hu')
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (0100q_9, 999, 4) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
astibm1*CLI
On simple run it is runs perfectly every time. Problem come when I am
using on a higher number of parallel calls. Because I am having a high
number of good log records in txts the problem must be somewhere else
probably in Asterisk variable handling for channels (par_telszam does
not overwritten with new value on new call file).
bye,
a
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Re: [asterisk-users] Automatic call-out problem
On Sat, Jan 19, 2008 at 01:53:47PM +0100, Artifex Maximus wrote:
Hello!
Thanks for your answer!
On Jan 19, 2008 12:49 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Jan 18, 2008 at 01:14:51PM +0100, Artifex Maximus wrote:
My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on
Fedora Core 4. I am making automatic call-out campaign with this setup
on 4 PRI. The scripts for this:
caller php script write this to outgoung folder:
fwrite($outfile,Channel: Zap/g1/$phonenumber\n);
fwrite($outfile,MaxRetries: 0\n);
fwrite($outfile,RetryTime: 5\n);
fwrite($outfile,WaitTime: 20\n);
fwrite($outfile,Context: 0100q\n);
fwrite($outfile,Callerid: $dbid\n);
fwrite($outfile,Extension: $phonenumber\n);
fwrite($outfile,Set: par_telszam=$phonenumber\n);
extensions.conf:
[0100q]
exten = _.,1,Wait(1)
exten = _.,n,Set(__TRIES=1)
exten = _.,n,Set(__FMT_DATE=%Y-%m-%d %H:%M:%S)
exten = _.,n,Set(__SZAM=${par_telszam})
exten = _.,n,System(echo -e
${SZAM}\,felvette\,${STRFTIME(${EPOCH},,${FMT_DATE})}
/tmp/0100q_0.txt)
What is this for? Why would you invent your own personal logging and not
use Asterisk's one?
We must know call is handled or not and which option callee choose for
a given question. It is an old script (back to * 1.0 era) and just
works till now. I only rewrite PHP script with Set in call file and
extension not using ${EXTEN} but ${par_telszam}. So the problem must
be there.
Use a unique-enough message and Verbose. What you use is quite
inefficient.
You are right but there is problem with Set through call file and that
is the question now. Grepping huge log file is efficient? Would be
great having persistent database connection in Asterisk and using
that. We had use MYSQL() but that was slow and complicated and finally
we switch to this kind of logging. Might not the best but working.
Anyway, why not show us a trace from the CLI?
set verbose 3
And see what happens when you drop a call file.
Here it is, I am masking out phone numbers:
astibm1*CLI set verbose 3
Verbosity is at least 3
-- Attempting call on Zap/g1/06x for [EMAIL PROTECTED]:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- Executing Wait(Zap/1-1, 1) in new stack
-- Executing Set(Zap/1-1, __TRIES=1) in new stack
-- Executing Set(Zap/1-1, __FMT_DATE=%Y-%m-%d %H:%M:%S) in new stack
-- Executing Set(Zap/1-1, __SZAM=06x) in new stack
-- Executing System(Zap/1-1, echo -e
06x,felvette,2008-01-19 13:16:33
/root/export/0100q_0.txt) in new stack
-- Executing Playback(Zap/1-1, 0100q_0) in new stack
-- Playing '0100q_0' (language 'hu')
-- Executing System(Zap/1-1, echo -e 06x,99,2008-01-19
13:17:10 /root/export/0100q_1v.txt) in new stack
-- Executing Set(Zap/1-1, TIMEOUT(response)=8) in new stack
-- Response timeout set to 8
-- Executing Set(Zap/1-1, TIMEOUT(digit)=1) in new stack
-- Digit timeout set to 1
-- Executing BackGround(Zap/1-1, 0100q_1) in new stack
-- Playing '0100q_1' (language 'hu')
== CDR updated on Zap/1-1
-- Executing System(Zap/1-1, echo -e 06x,1,2008-01-19
13:17:19 /root/export/0100q_1.txt) in new stack
-- Executing Goto(Zap/1-1, 0100q_2|999|1) in new stack
-- Goto (0100q_2,999,1)
-- Executing Wait(Zap/1-1, 1) in new stack
-- Executing BackGround(Zap/1-1, 0100q_2) in new stack
-- Playing '0100q_2' (language 'hu')
== CDR updated on Zap/1-1
-- Executing System(Zap/1-1, echo -e 06x,1,2008-01-19
13:17:25 /root/export/0100q_2.txt) in new stack
-- Executing Goto(Zap/1-1, 0100q_9|999|1) in new stack
-- Goto (0100q_9,999,1)
-- Executing Wait(Zap/1-1, 1) in new stack
-- Executing System(Zap/1-1, echo -e
06x,elkoszont,2008-01-19 13:17:26
/root/export/0100q_9.txt) in new stack
-- Executing Playback(Zap/1-1, 0100q_9) in new stack
-- Playing '0100q_9' (language 'hu')
-- Executing Hangup(Zap/1-1, ) in new stack
== Spawn extension (0100q_9, 999, 4) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
astibm1*CLI
On simple run it is runs perfectly every time. Problem come when I am
using on a higher number of parallel calls. Because I am having a high
number of good log records in txts the problem must be somewhere else
probably in Asterisk variable handling for channels (par_telszam does
not overwritten with new value on new call file).
Actually, even a log file as large as 1MB should be quite efficient to
grep. After the first time you read it is will reside in the cache for
the next times.
You can do some initial pre-processing (e.g: grep to a separate file)
if the log files are really huge.
A simple trace allows
Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
On Sat, Jan 19, 2008 at 06:21:15PM +1100, Rob Hillis wrote: I would suspect that your hardware is the cause of your problems. Running a production PBX system on a discarded desktop system is a /really/ bad idea. I would seriously consider an upgrade to your hardware. Well, there is not enough data to suggest that. Before blaming Ira for being such a cheap fellow (after all, he didn't buy one of those IBM big iorns to run Asterisk on) we should also consider that the upgrade to 1.4 probably also involved an upgrade of Zaptel, which *is* kernel space. And maybe there was soemthing completely different. Which is why I asked for a trace, to give some sort of direction to see where the problem comes from. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Polycom Provisioning Tool Released with BugFix
Michael Munger wrote: Polycom Provisioning Tool Updated. Looks like a Windows only tool. Shame it doesn't work under Wine. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Polycom Provisioning Tool Released with BugFix
I have reports to the contrary http://lists.digium.com/pipermail/asterisk-users/2007-October/199229.htm l Did you test it? Yours, Michael Munger, dCAP 404-438-2128 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Saturday, January 19, 2008 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] New Polycom Provisioning Tool Released with BugFix Michael Munger wrote: Polycom Provisioning Tool Updated. Looks like a Windows only tool. Shame it doesn't work under Wine. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Polycom Provisioning Tool Released with BugFix
Doug Lytle wrote: Michael Munger wrote: Polycom Provisioning Tool Updated. Looks like a Windows only tool. Shame it doesn't work under Wine. Doug Looks like it was written with VB.net. Not sure where Mono is as far as VB.net goes, but if I'm not mistaken, once its compile it should run on Mono. Try using MoMA to test for compatibility: http://www.mono-project.com/MoMA -- Warm Regards, Lee The only thing that kept me out college...was high school. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Polycom Provisioning Tool Released with BugFix
Lee Jenkins wrote: Doug Lytle wrote: Michael Munger wrote: Polycom Provisioning Tool Updated. Looks like a Windows only tool. Shame it doesn't work under Wine. Doug Looks like it was written with VB.net. Not sure where Mono is as far as VB.net goes, but if I'm not mistaken, once its compile it should run on Mono. Try using MoMA to test for compatibility: http://www.mono-project.com/MoMA Oops, looking a little closer, it appears that it is a standard win32 VB app and not vb.net. My mistake... -- Warm Regards, Lee The only thing that kept me out college...was high school. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
Tzafrir Cohen [EMAIL PROTECTED] writes: On Fri, Jan 18, 2008 at 12:20:56PM -0800, Ira wrote: Kernel panics can be caused by buggy kernel code and / or bad hardware. Buggy userspace should not (by definition) be able to cause them. If userspace can, it's a kernel bug. This is only true when userspace runs non-root. There are many many ways for root to crash the kernel. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nightly tarballs, would you use them?
I'd be even more likely to use nightly (or other periodic snapshot, even weekly) .deb packages. Because then I could use APT to notify me and manage them. Especially if they included a changelog (which APT reports), even if that changelog were only names of files/modules touched since the last one. On Sat, 2008-01-19 at 12:00 -0600, [EMAIL PROTECTED] wrote: Date: Sat, 19 Jan 2008 03:21:54 -0600 From: Russell Bryant [EMAIL PROTECTED] Subject: [asterisk-users] Nightly tarballs, would you use them? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Greetings, During the past week, there have been some requests for nightly tarballs to help making testing new Asterisk code easier. There was some debate as to whether they would be useful. The reason that they may not be useful is because you can get equivalent access to new code just by accessing the subversion repository directly. However, for one reason or another, some people would prefer to have a tarball. If this was available, would you be interested in it? If you just want to say yes or no for the sake of the poll, fell free to respond to me off-list. However, also fell free to respond here if you have more verbose comments on the topic that you would like to share. -- Russell Bryant -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Polycom Provisioning Tool Released with BugFix
Michael Munger wrote: I have reports to the contrary http://lists.digium.com/pipermail/asterisk-users/2007-October/199229.htm l Did you test it? Yes I did: Run-time error '339': Component 'Comdlg32.ocx' or one of it's dependencies not correctly registered: a file is missing or invalid Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] nokia e51
Hi all. Anyone to share some experience with Nokia E51 and asterisk? We are trying to connect the E51 to our asterisk but to no avail. Googling said that it should work, but we are seeing real strange things here: - tcpdump reveals the nokia is talking to other ports than 5060 - registration is not possible at all, right now there is no network traffic to the asterisk box at all. A softphone on the same wlan segment registers without any problem. The how-tos on the web suggest different settings concerning the proxy/registration setupBut none of them works for us. But we are not nokia guys at all So, any help greatly appreciated! The setup: Cisco AP with EAP-TLS. Connected to an switch on which several vlans are connected to a cisco router. The internal network (192.168.23.0/24) talks to the DMZ, on which the radius (for EAP-TLS) and also the asterisk box is hosted. IP Addresses are assigned via DHCP from the AP. The Laptop from which i am writing has x-lite installed and that works just fine with the same credentials we are trying to setup the nokia: 2001 abc sipgate No RFC3581 We have been playing with nat=yes|no, but we cant get it to work. Thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nightly tarballs, would you use them?
On Sat, Jan 19, 2008 at 03:51:43PM -0500, Matthew Rubenstein wrote: I'd be even more likely to use nightly (or other periodic snapshot, even weekly) .deb packages. Because then I could use APT to notify me and manage them. Especially if they included a changelog (which APT reports), even if that changelog were only names of files/modules touched since the last one. Binary packages are even more distro-specific. If you're interested in automating the build of a nightly deb yourself, I'd be happy to assist. We already do quite similar things for building asterisk from (packager's) svn. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIPAddHeader in .call file
Hi everyone, How can I add the equivalent of: exten = s,n,SIPAddHeader(Alert-Info: Ring Answer) in a .call file? This is to support paging to Polycom phones... Thanks for all info! Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPAddHeader in .call file
Sorry to answer my own post, but I have found a solution which perhaps
others can use too...
In the .call file, instead of specifying a channel line as:
chan: SIP/140 (for example)
use instead:
chan: Local/[EMAIL PROTECTED]
and put in extensions.conf
[polycom-paging]
exten = _1XX,1,NoOp(Paging Ext ${EXTEN})
exten = _1XX,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _1XX,n,Dial(SIP/${EXTEN},20,L(6))
exten = _1XX,n,Hangup
Steve Johnson wrote:
Hi everyone,
How can I add the equivalent of:
exten = s,n,SIPAddHeader(Alert-Info: Ring Answer)
in a .call file? This is to support paging to Polycom phones...
Thanks for all info!
Steve
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[asterisk-users] Debian packages (was: Re: Nightly tarballs, would you use them?)
Tzafrir Cohen wrote: On Sat, Jan 19, 2008 at 03:51:43PM -0500, Matthew Rubenstein wrote: I'd be even more likely to use nightly (or other periodic snapshot, even weekly) .deb packages. Because then I could use APT to notify me and manage them. Especially if they included a changelog (which APT reports), even if that changelog were only names of files/modules touched since the last one. Binary packages are even more distro-specific. If you're interested in automating the build of a nightly deb yourself, I'd be happy to assist. We already do quite similar things for building asterisk from (packager's) svn. I'd really love deb packages of the latest stable (1.4) tags of Asterisk in the official tree of Debian Etch. Regards, Philipp Kempgen ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
I wasn't intending to blame Ira for his own problems - I was intending to point out that running a production system on discarded hardware is a really bad idea. I wasn't even suggesting a mammoth server - as you may or may not have seen in my subsequent reply to him, the place I work for sells fairly low-end servers as Asterisk boxes which (at least in Australia) are comparable to mid to upper-mid range desktops in terms of pricing. 90% of the serious reliability problems I've seen are on hardware that people have taken the really cheap route on. Most people seem to think that Asterisk is a really cheap PBX. While Asterisk is certainly /cheaper/ than just about all comparable PBXs, if it's to be done properly and reliably it's certainly not dirt cheap. Evaluating Asterisk certainly can be since if it's only a test system, you can scrounge up some older hardware. The real mistake is in putting the older hardware into full production. Tzafrir Cohen wrote: On Sat, Jan 19, 2008 at 06:21:15PM +1100, Rob Hillis wrote: I would suspect that your hardware is the cause of your problems. Running a production PBX system on a discarded desktop system is a /really/ bad idea. I would seriously consider an upgrade to your hardware. Well, there is not enough data to suggest that. Before blaming Ira for being such a cheap fellow (after all, he didn't buy one of those IBM big iorns to run Asterisk on) we should also consider that the upgrade to 1.4 probably also involved an upgrade of Zaptel, which *is* kernel space. And maybe there was soemthing completely different. Which is why I asked for a trace, to give some sort of direction to see where the problem comes from. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calls Being Randomly Bridged
Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nightly tarballs, would you use them?
Matthew Rubenstein wrote: I'd be even more likely to use nightly (or other periodic snapshot, even weekly) .deb packages. Because then I could use APT to notify me and manage them. Especially if they included a changelog (which APT reports), even if that changelog were only names of files/modules touched since the last one. Have you tried the checkinstall app? It's a quick way to make a deb out of a tarball install. -- Russell ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
On Saturday 19 January 2008 20:32:42 Michael J. Liberatore wrote: Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. I have seen this exact problem when people park callers directly into numbered parking slots, instead of using the auto-distribution system. So, for example, the default distribution number is 700, and the parking slots are 701-720. Callers will get bridged if two callers are assigned to slot 701. This could happen even if only one person is doing the wrong thing -- one person uses 700 (correctly) and caller gets put into 701. Then another person transfers their caller to 701, and they're bridged. It comes down to a training issue. And yes, btw, you can use the CDRs to track down exactly who is doing the wrong thing. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R2-Unicall Asterisk as CPE and as CO
Hello Victor.
First, let me say I am confused about this:
I've changed the line (chan_unicall.c):
uc_callparm_calling_party_category(callparms,
UC_CALLER_CATEGORY_NATIONAL_SUBSCRIBER_CALL);
to
uc_callparm_calling_party_category(callparms,
UC_CALLER_CATEGORY_NATIONAL_PRIORITY_SUBSCRIBER_CALL);
because without this I cant receive calls from the telco. With or without
this I
can't place calls to the pbx.
I am quite sure you have made a mistake in this statement, why? simply
because this code is executed when YOU START the call to the far end
(whatever it is, Telmex or the other PBX), so it makes no sense to say
that w/o that change you can't receive calls, no sense at all. I am
sure you messed up somewhere else in the configuration files just like
possibly you are doing right now for the PBX. In anycase, I am about
to make a new release of chan_unicall Asterisk driver that will
include a way to modify the calling party category from the dialplan
extensions.conf
Now, regarding your problem when receiving calls from the pbx, I think
you have configured the PBX to not send ANI digits, and you configured
chan_unicall to expect ANI digits, hence the timeout. Try configuring
Asterisk with 0 callerid for the PBX side, or configure the other PBX
to send the proper number of ANI digits.
Regards,
Moises Silva
On Jan 18, 2008 9:41 AM, Victor Toofic [EMAIL PROTECTED] wrote:
Hi!
Im having some troubles trying to configure * as a bridge between a telco
and a pbx with R2, the scenario is this:
E1/R2-E1/R2
| Telco |-| * |-| PBX|
| (Telmex) | - | |
I can receive calls from the telco and can place calls to the pbx, I also
can place calls to the telco.. but I can't receive any calls from the pbx.
When receive a call from the pbx I get this:
cause 32771 - T3 timed out
If I connect the pbx directly to the telco there is no problem, the calls
are stablished correctly.
Im using the package at:
http://www.moythreads.com/astunicall/downloads/
http://www.moythreads.com/astunicall/files/astunicall-1.2.25-0.1.tar.gz
that contains:
asterisk-1.2.25
spandsp-0.0.4
unicall-0.0.5pre1
libmfcr2-0.0.3
libsupertone-0.0.2
libunicall-0.0.3
zaptel-1.2.22
My zaptel.conf is this:
loadzone=mx
defaultzone=mx
span=1,1,0,cas,hdb3
span=2,1,0,cas,hdb3
span=3,0,0,cas,hdb3
span=4,0,0,cas,hdb3
cas=1-15:1101
cas=17-31:1101
cas=32-46:1101
cas=48-62:1101
cas=63-77:1101
cas=79-93:1101
cas=94-103:1101
cas=110-124:1101
and unicall.conf is this:
[channels]
usecallerid=no
hidecallerid=no
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=no
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
immediate=yes
loglevel=255
protocolclass=mfcr2
protocolvariant=mx,10,4,16
group=1
protocolend=cpe
context=incoming1
channel = 1-15
channel = 17-31
group=2
protocolend=cpe
context=incoming2
channel = 32-46
channel = 48-62
protocolvariant=mx,10,8
group=3
immediate=yes
usecallerid=yes
protocolend=co
context=incoming3
channel = 63-77
channel = 79-93
group=4
protocolend=co
context=incoming4
channel = 94-103
channel = 110-124
The port #1 of a TE405P card is connected to the telco and the port #3 is
connected to the pbx.
I've changed the line (chan_unicall.c):
uc_callparm_calling_party_category(callparms,
UC_CALLER_CATEGORY_NATIONAL_SUBSCRIBER_CALL);
to
uc_callparm_calling_party_category(callparms,
UC_CALLER_CATEGORY_NATIONAL_PRIORITY_SUBSCRIBER_CALL);
because without this I cant receive calls from the telco. With or without
this I
can't place calls to the pbx.
When I receive a call from the telco I place it directly to the pbx.. and
that works ok:
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 - 0001
[1/IDLE/Idle /Idle ]
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 Detected
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 Creating a
new call with CRN 32770
Jan 16 12:27:01 DEBUG[4136] chan_unicall.c: MFC/R2 UniCall/2 1101 -
[2/DETECTED/Seize ack /Seize ack]
Jan 16 12:27:01 NOTICE[4136] chan_unicall.c: Unicall/2 event Detected
Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
At 04:44 AM 1/19/2008, you wrote: Well, it's been very reliable. It's been running 24/7 for 2 years and the only problems have been my putting bugs in the dial plan, problems with SIP providers going broke and trying 1.4. So how exactly would more expensive hardware have improved my reliability? I really wanted it to be reliable. I ran a duplicate network for the phones so they don't share bandwidth, I bought good phones. Everything is POE with the POE switch, the Asterisk box and all the networking stuff to the outside on a big UPS. Do you actually think the odds of a HP desktop sitting on a UPS in a cool corner doing nothing suddenly dying are much greater than a bottom end server box doing the same? It seems to me unless I want to go dual PS and flash drives that I'm not going to do much better than I have now. Ira What you run it on is very much a function of how reliable you want the system to be. The better the hardware, the more reliable it will be. If you're running in a business environment, then I wouldn't recommend anything less than server grade - even if it's low end server grade. The company I work for supplies either Dell PowerEdge 860s (1RU servers that are similar in price to an upper mid-range desktop) or Dell PowerEdge 840s (tower cases that are similar in price to a mid-range desktop) Running on cheap hardware is a great way to cost yourself more in the long run - in lost productivity, lost sales and IT support. Except that it's been running 1.2 for 2 years with no problems. It has to handle all of about 20 to 30 calls/day so it's not really under much load and I see no reason why it should be a problem. If not for the TDM404 I would probably try to put it on a NSLU2. What would you recommend I run it on? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Being Randomly Bridged
On Sat, Jan 19, 2008 at 09:32:42PM -0500, Michael J. Liberatore wrote: Hi i have a friend who i setup an asterisk system for at his doctors office. it has 3 snom 360 phones with 6.2.x stable firmware and latest asterisk 1.4 and zaptel. They have the digium 4 port fxo card. They are extremely upset because calls are being randomly bridged for no rhyme or reason. They say that callers will call in and sometimes get connected with other callers, or they will be in the queue and then be talking to another caller waiting in the queue or on hold. Or they will be talking to a patient and then have another patient end up on the conversation. They are freaking out because of hippa and laws that govern privacy but i have no clue why. I assume most cases are conference calls being initiated by accident. So any help would be greaat. maybe just disabling conference calls would be a good start but i dont know how with sip phones. or maybe this is a bug? unfortuinately they dont give me much info and i dont use the phones so i dont have any specific logs to show, they just call me freaking out saying this stuff but they rarely can give me a specific call cause they get so many. Can you provide a more detailed trace of such an event? (Use more verbose logging, and such) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
PC's age and when they age, things tend to go wrong, particularly when you upgrade software. Unusual crashes are usually the first sign that something is going wrong. To me, it sounds like you've put the money into many of the right areas - segregating your voice and data networks, going with decent phones and ensuring your power is reliable so it just seems a little strange to go cheap on the actual server. It sounds like you've been pretty lucky with this machine - not all desktop machines are going to be anywhere near that reliable. The big thing that server grade machines give you is better quality of parts that have been extensively tested with a range of operating systems. I guess it's up to you - personally, I'd take the warning signs and start planning to replace the server. Possibly I'm just a little more cautious than some. :) Ira wrote: At 04:44 AM 1/19/2008, you wrote: Well, it's been very reliable. It's been running 24/7 for 2 years and the only problems have been my putting bugs in the dial plan, problems with SIP providers going broke and trying 1.4. So how exactly would more expensive hardware have improved my reliability? I really wanted it to be reliable. I ran a duplicate network for the phones so they don't share bandwidth, I bought good phones. Everything is POE with the POE switch, the Asterisk box and all the networking stuff to the outside on a big UPS. Do you actually think the odds of a HP desktop sitting on a UPS in a cool corner doing nothing suddenly dying are much greater than a bottom end server box doing the same? It seems to me unless I want to go dual PS and flash drives that I'm not going to do much better than I have now. Ira What you run it on is very much a function of how reliable you want the system to be. The better the hardware, the more reliable it will be. If you're running in a business environment, then I wouldn't recommend anything less than server grade - even if it's low end server grade. The company I work for supplies either Dell PowerEdge 860s (1RU servers that are similar in price to an upper mid-range desktop) or Dell PowerEdge 840s (tower cases that are similar in price to a mid-range desktop) Running on cheap hardware is a great way to cost yourself more in the long run - in lost productivity, lost sales and IT support. Except that it's been running 1.2 for 2 years with no problems. It has to handle all of about 20 to 30 calls/day so it's not really under much load and I see no reason why it should be a problem. If not for the TDM404 I would probably try to put it on a NSLU2. What would you recommend I run it on? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-1.2.26.tar.gz Thoughts?
On Sun, Jan 20, 2008 at 06:33:31PM +1100, Rob Hillis wrote: PC's age and when they age, things tend to go wrong, particularly when you upgrade software. Unusual crashes are usually the first sign that something is going wrong. And suddenly the same PC has unaged when reverting to 1.2? Again, you don't have enough data to be conclusive on that. So I humbly suggest that you won't be. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
