parallel calls.
jg
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://www.sangoma.com/products/d500-400-2000-sessions/ or
http://www.sangoma.com/products/d150-30-400-sessions/) and examples on
how to use them.
jg
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Hi!
You could do a sequence of Dial cmds and dispatch the various hangup
causes. If a device is busy, the time delay is pretty short. Then you
can decide whether inviting another device makes any sense.
jg
Am 15.04.2013 11:08, schrieb Santi Anton:
Hi,
Can a call to multiple devices
of Background.
jg
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But you do have a cdr with disposition NO ANSWER, do you?
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jg
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It looks like your task has the same basic requirements. Setting up a
call file based system is not very difficult, but details like
pronunciation of the guest's or patient's name may involve some
additional work.
jg
Hans,
they are currently calling patients. I think these calls apply only to a
certain fraction of the patients, who are difficult to contact by other
methods.
jg
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being, but the impact for the health center could be a lot less,
because they don't have to dial anything or wait for a call to get answered.
jg
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... Essentially, I suggested a predictive dialer
(http://en.wikipedia.org/wiki/Predictive_dialer). In this case this
could be a reasonable thing to do.
jg
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Here are your answers:
1st question: Anything that makes sense.
2nd question: Maybe
Please, explain your setup.
jg
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Read it (http://the-asterisk-book.com/1.6/minimale-telefonanlage.html),
or regret it!
jg
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Maybe using a LOCAL channel could help. One ext. for Snom with Snom
header, another for Digium with Digium header, then simultaneously call
both local channels, which then call the appropriate phones.
jg
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HWEC, then you do get the default
software canceler. Depending on your needs, switching to the OSLEC
canceler might or might not be a better choice. Generally, a card with
HWEC gives the best results, especially when you also have a lot of
facsimile messages.
jg
No, no. Wanpipe is more or less the low level interface to Sangoma
cards. They interface with Asterisk using the DAHDI package as the
Digium cards do.
jg
Am 17.05.2013 16:59, schrieb Angelo Delphini:
Look ...
wanpipe sangoma only to cards Digium is dahdi.
Blz
for the RTP streams or are the SIP phones responsible?
jg
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a Sangoma B700 hybrid card. You do need an adapter (if it
doesn't come with the board or barebone) as the cards are at least of
2-U size.
jg
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You need to download libsrtp manually from
http://srtp.sourceforge.net/srtp-1.4.2.tgz
I never had any problems compiling this packages.
jg
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Your problem looks more like an MTA configuration problem. You need at
least a valid relay host.
jg
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Sangoma's tech support is probably the better source of information.
DAHDI: obviously DAHDI channel
i: incoming call
3: span 3 (not the port)
211123456: CLID, probably subject to filtering (see
national/international prefix settings)
89c: internal counter (i.e. 2204 calls so far)
jg
.
For a BRI device a single span has 2 channels, a PRI device up to 30. As
far as channel variables go the actual channel does not seem to get
reported, but this is not really necessary.
jg
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calls coming in and out:
- keep a meaningful pri show spans status (pri show spans outputs Down
when the line is down (cable unplugged, no q921 traffic, ...))
- avoid sporadic ERROR messages in logs.
Did you have any problems with the line status?
jg
see a SABME frame before a SETUP request.
jg
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You could start with a standard installation. Once things are running
you can specify unneeded modules in modules.conf.
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-5-SECT-1.html
explains how things are related.
jg
. In environments where callers
are announced to C, C would typically not want to wait for A---believe me.
jg
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circumventing Asterisk.
jg
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phone has intercom capabilities, this might be a
quick and dirty solution that works.
On the other hand, if B does the transfer and knows about the state of
the transfer, why should there be an extra announcement?
jg
How about
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DB.html
?
jg
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Do you see the channel driver chan_dahdi.so in /usr/lib/asterisk/modules?
If not, go back to the * src dir, issue a ./configure, then make make
install and check what * got this time.
If you have played with menuselect you might have to check these
settings, too.
jg
.
jg
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facsimiles in context [fax-out]
[fax-out]
exten = _X.,1,Verbose(1,Outgoing fax...)
same = n,Dial(...${FILTER(0-9,${EXTEN})},40)
same = n,Hangup()
Email sending is configured inside Hylafax.
jg
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Enable logging and see what happens when you start and stop the iaxmodem. Obviously, it doesn't
register, but the messages might be helpful. Since iaxmodem is somewhat older, you might have to
disallow call tokens in iax.conf.
jg
If I read your log entries correctly, you are not playing any MOH at all. BackGround() normally
plays sound files from the language dependent sound directory.
jg
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for the transfer, so that Asterisk might already not know
anything about D. Except for C, everybody is inside the same subnet.
jg
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Did you specify a timeout value?
jg
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More things to try:
(1) Is there any entry in /var/log/maillog (or equivalent log file)? If so, mutt basically works
and the messages should give some clues.
(2) What happens if you call mutt without any attachments?
I am using mutt in exactly the same way and it works.
jg
Am 19.06.2013 21
Have you checked whether the same codecs, or codecs with the same bandwidth
requirements, are used?
jg
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and whether there are any general ways to control this size.
jg
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Aha. I overlooked that some phones had ulaw:10 in sip.conf, instead of the standard ulaw:20.
That explains the packet count difference. It seems my call quality issues are coming from
something else.
... and this explains how to set the packet size. Answer to get answers, or so.
jg
, but
this shouldn't matter). I'll have a look at your log entries tomorrow.
jg
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Have a look at the documentation of the channel bank. I guess some kind of overlap dialing is
enabled, which is typically associated with a timeout value. chan_dahdi.conf also has entries
like this.
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go away I could tell next week what my system is doing.
jg
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Are you using raw AMI or AMI via HTTP?
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/d9e5cb9b-c5fa-429b-9974-a7b4f4bf58c3.html). The tool is now outdated
(partial DAHDI support, Woomera obsolete), but the SIP channels are usable. If you have a few
SIP channels you could compare the behavior of both packages under load.
jg
there is plenty of room for wear leveling
optimizations.
jg
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When you have many calls, there are usually (read/write=all) a lot of RTP, RTCP, and VarSet
events. This might slow down things, but whether they occur or not depends on your configuration.
This might be another thing to look at.
jg
I guess this was a question for Alexander. As far as I am concerned, I never had such a load
that slowed down AMI event processing (responses within at most 1/10 of a second), but for
future tests I should probably set up a real torture test.
For a robust PBX application, it would make sense
about the current models.
jg
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Maybe this is a stupid question. Are the files in Extra Sound Packages related to any product
or are they just supplemental material? I searched the source files for some of the file names
and didn't find any reference.
jg
configure your phones to use a different ring tone for each account. In addition you would also
have some visual feedback (line buttons, display text).
jg
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You can use anything that is discriminable to connect a call to its proper endpoint. SIP headers
have the disadvantage that the format may depend on a the phone model.
jg
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You probably picked the wrong channel.
There are two bridged channels with slightly different unique ids, which are actually
timestamps. You should watch for the Bridge event that ties the two channels up. The Bridge
event contains both channels, both UniqueIDs, and both CallerIDs.
jg
phones
have on-board.
jg
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I just got access to an older Asterisk 1.6.2.18 box and found that the multiple transfer problem
does not exist here. So with 1.6.2.18 I can transfer as often as I wish using DTMF sequences.
jg
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Well, I forgot to add the t or T option to the dial command, which is required to do transfers
with DTMF sequences.
jg
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What is the value of charset in voicemail.conf?
Have you tried a different Email client?
jg
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If I read your mailcmd correctly you are not really mailing but just dumping the data. Is the
display correct when you use the default setting /usr/sbin/sendmail -t? You could send the
mail to a local account and open it with mutt.
jg
Some time ago I had a similar problem but it turned out to be a display problem of the email
client (Outlook Express on an old XP machine).
jg
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I checked the raw text of my voicemail messages today and I saw pretty much the same escape
sequences for UTF-8 like you did, but I do not have any display problem. You could save the
message locally and hand edit it (starting with uppercase UTF instead of lowercase utf).
jg
I checked your original message, and I guess the expected string was a little
bit different:
1504|12|Teste - Rafael 1570|0:16
I can't see anything wrong with quoted printable decoding. My best guess is still the email
client and its settings.
jg
How about ${EXTEN:-1:1}?
The Definitive Guide has a special paragraph with the title *More Advanced Digit
Manipulation.*
jg
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Damn it!
Try: ${EXTEN:0:-1}
jg
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You do not need to calculate the jitter values yourself. For a quick check you can use the CLI
cmd sip show channelstats. For external monitoring you could capture the RTCP AMI events.
jg
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there must be something that sets this timeout value.
jg
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, but the the Snom phone returned Got SIP response 486 Busy Here back from
Now, I do think it is a phone setting and it might be hard coded.
jg
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been made.
Does anybody know what to do with these messages?
The BRI cards are 100% ok and I have never seen these messages when connected
to a public circuit.
jg
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Q.921 just says to log the condition and carry on as if it did not happen.
I guess you mean: carry on if it happened (and don't care any more).
Sorry, English is not my mother tongue. I didn't see the 'not'.
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jg
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Does your Dial() command include the m option?
jg
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you do get the
invitation, everything is fine.
If you really can't remove the m, you could still use an audio file with a funny ringtone and
stuff this into an moh class. Dirty, but it will work.
jg
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Have you considered using VoiceMailMain()?
jg
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If you use a standard command line mailer (like mail, or mutt), then you can simply concatenate
the email addresses. You could do something like this:
[globals]
...
EMAIL=mdiehlena...@gmail.com,a...@nowhere.com,d...@nowhere.com,
...
same = n(sendit),System(echo ${msg} | mutt -s blabla ${EMAIL})
Well, my suggestions is not so good with Comedian Mail.
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Is there a special reason why you do not evaluate the CDRs? The Call Detail Records would answer
your questions and you could even add custom fields.
jg
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possible side effects if Local channels are used explicitly. This would require adding a
persistent channel variable (the ones with __).
I apologize if this type of question has already been asked before.
jg
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create a separate context for outbound calls.
Wouldn't that be more or less identical to my way? I would have to dispatch the channel to see
whether it is allowed to enter the outbound context. Maybe I misunderstood something.
jg
of this potential problem, or am I not aware of some
important concept?
jg
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in features.conf can be
deleted).
jg
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that they are not as stable as one would wish, at least at a time scale of several hours.
When using several moh file classes, I have never hat audio problems.
jg
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My CPU load is permanent at 200-250%. I have 7 active mpg123 streams.
I forgot something. Even if your 7 streams are mp3 streams they cannot consume the CPU power you
are seeing.
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is probably a safer thing to do than
just killing mgp123.
My system issues an moh reload maybe once or twice a day, and sometimes not even a single
time. When I use Shoutcast streams there a a lot more restarts a day.
jg
() is actually doing.
jg
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systems. For a recent Fedora based test system running under VirtualBox
I get typically varying values as low as 550 for timing test 1024, but this does not seem to
have any major influence on plain calls.
jg
Not that I know. You could monitor the log file and generate a UserEvent (call
file or AMI command).
jg
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.
jg
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Looks like a perfect example for rights management using differentiated contexts. The Asterisk
book has some sample code. From the perspective of Asterisk it doesn't matter if you handle the
rights for outside calls (national, international, ...) or just for some internal calls.
jg
data. Just
search for adaptive ODBC, or read the Asterisk book.
jg
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something else happens.
jg
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: Goodbye
ActionID: 4711
Message: Thanks for all the fish.
jg
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Hmmm, do I understand you correctly that you park and unpark a call using the
same phone?
If yes, why does simply holding the call does not work? The SPA504 has an extra large button
on the right for this and you don't need any support in the dialplan.
jg
As I said:
- Do it yourself (Action Logoff)
- Process the Shutdown event
- control the state of the socket
I have Manager sessions running for hours without any connection losses. So if you have
connection losses then there is likely something else.
jg
Can you describe your problem?
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beyond Asterisk events.
jg
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(typically from-internal). You also need to set the transfer options properly,
or this could be abused.
jg
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. Of course a pure VoIP system does not need to consider this.
jg
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Does it matter? I thought keys are case insensitive.
jg
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table. This avoids any string comparisons in the rest of the program.
jg
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can get away by duplicating the classes for
the Link and Unlink events and give them proper names.
jg
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There are several options: AMI and ARI (vs. 12). Depending on what you are trying to do there is
also AGI and FastAGI.
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telco (or they pick a default number). There are exceptions, but not for mere mortals.
jg
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