I am not sure about the problem but note that it may be related to incorrect
IP being used. Sometimes, WAN 1 and sometimes WAN 2
Most likely. Get a provider that uses IP authentication rather than
registrations, and enable access from both of your WAN IPs. All set.
Luki
Unfortunately, if I kill all asterisk-processes with kill -9 ..., a
coredump never is writen to /tmp, I also looked in other dirs.
Try kill -6 (i.e. SIGABRT). That usually triggers a core dump for me.
Luki
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However, when I connect a PC to that port, SPA922 works as bridge.
Exactly. The SPA9x2 has a 2-port switch; no NAT, no routing (unlike
the SPA2102, etc).
I think the 5.1 series is the latest firmware for the 922; the the
942, there is 6.1.5a.
Luki
Are there any CLI commands to free this up or any other ways without having
to restart asterisk.
Did you try soft hangup channel? Or set an RTP timeout to avoid
abandoned channels?
Luki
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).
This particular box only handles signaling from a dozen static peers.
No registration, no media (directrtpsetup=yes), no NAT, no
transcoding, no MOH... but it does use realtime for SIP and IAX, and
AGI and DeadAGI for routing.
Luki
DeadAGI on a live channel will cause problems, please use AGI
The good news is, we run tens of thousands of calls every day through
this box and about half of them spit out this warning, but it never
caused any problems for over a year. Thus this warning is probably
safe to ignore.
Luki
by now.
Luki
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asterisk-users mailing
should only do it for the first caller.
MeetMeCount() will help. The caveat is that if the first caller
disconnects, the remainder of the conference will not be recorded.
If anyone has a better solution, please tell us :).
Luki
(__TARGET=${EXTEN})
Dial(SIP/phone1Local/pho...@common_area)
[common_area]
exten = _phone.,1,Set(CALLERID(name)=${TARGET}: ${CALLERID(name)})
exten = _phone.,n,Dial(SIP/${EXTEN})
Something like that. I hope I got all the () and {} right, I don't do
that much dial-plan coding anymore...
Luki
to that location. However, I don't
think you will gain much.
Luki
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it's an
authentication mismatch between the matched peer and the peer name in
the SIP message. Try turning sip debug on and see if the packets give
you some hints. Incoming calls also always work for me.
Luki
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else
would run. Kernel would respond to pings, but that's it. We no longer
use realtime priority for that reson :).
Luki
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the profile and recreate
it.
Luki
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selects is different with every
dialog, so that doesn't help either.
Any input would be appreciated before I throw that phone out of the window.
Thanks,
Luki
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Darryl,
OK, that could work but it makes the use of these phones behind
consumer routers rather impossible. How many of those will inspect and
transform SIP packets? Oh why does Cisco have to do things differently
from everyone else...
Luki
2009/11/16 Darryl Dunkin ddun...@netos.net:
You need
Martin,
sounds like the hiccup my E71 had once. I think the symptoms were
identical. Changing the transport type from Auto to UDP solved the
problem for me. The Auto setting worked, but only sometimes. Maybe the
E65 is similar...
Luki
2009/9/12 martin f krafft madd...@madduck.net:
Hey folks
city. Say in Hamburg, Germany, your number can be as short as 5
digits or as long as 10. You really have no way of knowing.
Luki
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rate for an intrastate
call if the CallerID is set out of state, but IMO that doesn't make a
good impression and isn't worth the savings. YMMV.
/Luki
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conflicts by rewriting the
ports transparently.
Bottom line, a few phones behind a well-behaved NAT should work just fine.
/Luki
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. The actual VoIP
service is excellent; billing and paperwork can be messy at times.
Luki
On Dec 15, 2007 4:25 PM, Steve Finkelstein [EMAIL PROTECTED] wrote:
Hi all,
There's a myriad of options these days and I haven't been keeping up to date
with what's respectable any longer.
I essentially need
. The packets are being dropped for whatever
reason and never reach the asterisk process. Check your iptables and
RTP port range, and perhaps try changing it.
Luki
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. But I'm not complaining... just don't
have a better idea how to fix it.
Luki
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with the media at all, but if you let SER handle
registrations and authentication, then I'd rather not keep track of
codecs/DTMF on asterisk as well.
Those two have been bugging me most.
Luki
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Luki, thanks for writing to say it DOES work. I've have just now had
another look, found my mistakes (basically $MAC instead of $MA), and
it's working!
I'm glad you got it sorted out. Yes, it works with XML or compiled
files. To help with troubleshooting, specify a syslog server and set
works just fine and very reliably. We have disabled periodic
resync as the Sipura phones seem to reboot sometimes for no good
reason when they apply the new but unchanged profile. If there is a
config change, we just push it on the phone with SIP NOTIFY option.
--Luki
Is it possible to configure a PAP2 to
auto-answer for either paging or intercom?
No. You cannot force the connected device (phone) to auto-answer.
Imagine you have a plain old phone attached to it, who's going to lift
the receiver?
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is
somewhere preset in the code. If you do NOT use the r flag, asterisk
simply passes call progress indications from the source, without the
need to generate any. Hence no error, and you hear ringing.
Yes, it's a bug, but there no magic in the symptoms you observe.
Luki
potential for
congestion, etc. Of course with the Internet being a best-effort
network there are no guarantees, but by minimizing the potential for
trouble you can achieve decent quality nevertheless. So, try to find a
provider near you focusing on your market.
Luki
into something like that. And you can expand it fairly
easily by adding another SPA for a second line.
Luki
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I did find out how to add the sip message for distinctive ring
i just dont know what variable needs to be passed in
order for it to work.
Try: SetVar(_ALERT_INFO=Bellcore-r2);
etc.
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, samba or bind of 200+ days. I have therefore reasons to believe
that the hardware is OK.
So go figure. And BTW, the crashes (based on the core dumps) are
always at a different place. There is no consistency. Right now I'm
just glad it no longer core dump on me :).
--Luki
offline long enough for a memtest?
The machine has been retired (routine upgrade cycle). But I hardly
doubt that was the problem. My guess is it was somehow related to
limited CPU power (thread switching, interrupts, or whatnot). The old
hardware was single CPU and a lot slower.
--Luki
I don't know how to contact them, but I am having the same problem.
The domain is registered to Jed Stafford. If you want the domain contact
details you can do a whois.
The same Jed Stafford as SellVoip.net? Oh yes, that explains it... see archives.
--Luki
pings and
traceroute look like with 15 calls up. Could you try SIP instead of
IAX? Sounds like the problem might be upsteam from you.
--Luki
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*think* only the exact 1-900 prefix is a premium rate call.
--Luki
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as good as g711 (a bit
duller), music is acceptable. Transcoding overhead is low, many ATAs
support it, and the bandwidth (with overhead) is about 40 kbit/sec.
It's a good alternative, IMO.
--Luki
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:
Average CPU utilization per call: 0.137% (~17 MHz)
--Luki
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it itself.
--Luki
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(). But it's rather beta.
--Luki
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guaranteed that that there is someone to answer the fax, given
sufficient resources (CPU and memory). The biggest drawback with
app_rxfax is that if it crashes for whatever reason (happens
sometimes), it will take down the entire PBX and all sessions with it.
--Luki
for no apparent reason. I had more trouble when
trying to use T.38 with the newest app_rxfax so I abandoned it for
now. And iaxmodem cannot do T.38 anyway...
So you are saying a pool if iaxmodems and a loop through Dial() to
find an open one is the way to go?
--Luki
PROTECTED])
exten = 1,n,Hangup
exten = 1,n(good_timing),Dial(SIP/techsupport)
You get the idea... but I agree, why on earth would you want to do
that? We only provide 6.7% tech support?!
--Luki
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,2,Dial(SIP/[EMAIL PROTECTED])
--Luki
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there. And yes, I did reboot
the Sipura.
--Luki
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Obviously somewhere in the asterisk code 30ms must be coded... is it set in
just one place, and if so can I set that to 20ms?
The default is 20 ms for most (all?) codecs. It's in rtp.c, where
ast_rtp_write() creates a new smoother.
--Luki
.
Something strange is going here, and my bet is on some kind of NAT
screw-up.
--Luki
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Just how many SIP packets do you think it takes to set up a call?
Probably around 8 - 10 per call, excluding any ReINVITES, DTMF, etc.
INVITE, Authentication Required, ACK
INVITE w/AUTH INFO, TRYING, RINGING, OK
BYE, OK
--Luki
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.
See if that helps.
--Luki
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: Never tried it...
--Luki
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two contexts and no macro, etc.
--Luki
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, you're looking at
about 90 call setups/tear downs a second.
I don't think even without running the RTP through Asterisk this box
could handle 10922 concurrent calls.
--Luki
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putting a Ringing(); before the dial statement to let
the SIP phone know the call is being connected. I believe once
progress comes from the Dial command, it will replace the Ringing.
However, if your channel bank answers the call right away, this won't
help.
--Luki
mirroring to sniff the traffic.
Good luck and keep us posted.
--Luki
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No such host: 3213)
Look for an extra closing parenthesis in your Dial command:
Dial(SIP/3210-084eaa80, SIP/3213)|30|to)
It should be SIP/3213 rather than SIP/3213).
--Luki
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)
That will send the callerID number followed by a new line. You can of
course change the format to your desire. Make sure /dev/ttyS0 is
writable by the asterisk user, and is also properly set up (baud rate,
bits, ...).
--Luki
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on a single
PIII-600, 256 MB RAM. CentOS 4.4 with a stock 2.6.9-42 kernel.
Asterisk 1.2.5, in production for 1.5+ years. CPU usage about 2% per
call. Quite reliable (hence not upgraded). This is a g711 only setup
with no transcoding.
--Luki
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guess adding 0.1 Mbps for call setup and tear down is safe.
--Luki
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degraded. I guess I
spoiled them with ulaw. So no g729 here. g726-32 on the other hand was
acceptable, although the difference is still noticeable.
--Luki
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protocol traffic.
--Luki
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exit
with Ctrl+C -- the mySQL client does it.
--Luki
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Luki, thanks for the response. Could you give me an
example of the use of vmauthenticate in a very short
dialplan?
Thanks
Jez
*CLI
-= Info about application 'VMAuthenticate' =-
[Synopsis]
Authenticate with Voicemail passwords
[Description]
VMAuthenticate([EMAIL PROTECTED]|options
\n
voicemail.conf.\n
If the mailbox is specified, only that mailbox's password will be
considered\n
valid. If the mailbox is not specified, the channel variable
AUTH_MAILBOX will\n
be set with the authenticated mailbox.\n\n
Options:\n
s - Skip playing the initial prompts.\n;
--Luki
thttpd or qmail for 12+ months without a hiccup. So
it's hard to blame it on the hardware because no other program seems
to crash randomly on the same machine.
--Luki
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exten = context,1,Dial( SIP/[EMAIL PROTECTED])
exten = context,2,Dial(SIP/[EMAIL PROTECTED])
Currently, if the first number doesn't answer, the session is closed.
Specify a time out. Without it * will not continue to priority 2 if
[EMAIL PROTECTED] is reachable but does not answer.
exten =
other them.
T.38 pass-through support in Asterisk is available as a patch on the
bug tracker for 1.2. Not sure if it made it into the 1.4 beta version
or not, but on 1.2 it works OK for me.
--Luki
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] -- Progress
[Sep 21 17:49:48] -- Peer audio RTP is at port 1.2.3.4:12345
etc.
--Luki
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of the time) but having
it receive faxes with T.38 would be ideal.
Can this be done already?
--Luki
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definition). That's at least my interpretation.
--Luki
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4096 Aug 7 17:44 tmp
-rw-r- 1 asterisk asterisk 264826 Apr 9 15:53 unavail.wav
--Luki
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and it worked quite nicely.
This was before 1.2 so it may no longer work at all. At the very least
it will likely required some updating. Doable, just depends how much
time you want to put into it :).
See: http://bugs.digium.com/view.php?id=4371
--Luki
a directory where core files should be dumped with:
mkdir /corefiles
echo /corefiles/core /proc/sys/kernel/core_pattern
The kernel will then dump all core files for any process into the
/corefiles directory.
--Luki
On 7/25/06, Patrick Cervicek [EMAIL PROTECTED] wrote:
To get a core file, I
Thameem,
0180's are special. Some are billed per connection, some per minute.
Typically the higher the next digit the more expensive it is. 0180 1
is same a local call from anywhere in Germany. See:
http://www.elektronik-kompendium.de/sites/kom/0312221.htm
--Luki
with SPA 1000, 1001 and Grandstream.
Interesting... any ideas what the heck is up with that? This is
software version 3.1.9(LSa). I can't upgrade the software because the
unit thinks it's not idle and hence does not start the upgrade
process. Kind of disappointing.
--Luki
it's DOA. But more than one?
--Luki
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over from there
and place the call.
See:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Make sure you create the file elsewhere and move it into the
directory, and that it is readable by asterisk.
Luki
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]
10. SayDigits(${R}) [pbx_config]
-= 1 extension (10 priorities) in 1 context. =-
--Luki
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is connected. I may be
wrong so just try it :). The ATA must be able to talk directly to your
provider in such a case (i.e. not NAT).
--Luki
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only a router) so if you need more info let
me know.
--Luki
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reasons).
--Luki
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Who is the file who listen when a softphone is run from a remote pc?
-- Registered SIP '651' at 192.168.251.10 port 2209 expires 900
chan_sip.c
--Luki
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. The
detector in the Sipura is too sensitive but you can avoid it by not
using INFO or INFO+INBAND... just pure INBAND. Seems to help in the
1000, 1001 and 2000. Can't comment on the 2002.
--Luki
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outgoing. Anyone else using good service
providers they can recommend?
That's something to the -biz list, probably but you may contact me off
list if you need suggestions.
--Luki
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can we enable or force a codec on specified npa..
Depends on the channel. On SIP you can set SIP_CODEC to force a codec,
but I don't think you can disallow one in the dialplan.
See:
http://voip-info.org/tiki-pagehistory.php?page=Asterisk+variables
--Luki
the answer.
--Luki
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a need, I guess Eric can
explain it further...
--Luki
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Why don't you put it up somewhere, if you need space I can put it on
tel.net ?
Yes, putting it up for download somewhere would be nice. I'd be
interested too and I certainly can provide space for it too ...
although that doesn't seem to be an issue, I see.
--Luki
traffic will flow between the organization and the provider.
But each organization needs to be reachable via a public IP (i.e. not
NAT) and the provider you use must support it too. I believe most(?)
do, but you should check.
--Luki
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copies over --
ldd is your friend.
--Luki
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in the
chrooted environment as well:
chroot /usr/local/asterisk license-installation-script
I don't have time to write up the steps to chroot asterisk, but if
anyone is interested then I will tonight.
--Luki
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the timestamp to the current time. Give it
a shot.
Luki
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working right (including trying 40 ms packets to reduce the number of
packets/sec; our modem was choking with 500 outgoing packets/sec),
etc.
It can be done. It all depends how much risk you are willing to take,
and how important setup and operating costs are to you.
Luki
is the advantage compared
to having 4 GB (dedicated) RAM in the machine and making a RAM disk
with it? You need the RAM either way and that ought to be at least as
fast as this card on a 33 MHz PCI bus. You loose the non-volatile
advantage but that's about it, no?
--Luki
more than a year to collect all the
responses... and who says the credentials do not changed periodically
and the ATA fetches new config from Vonage?
--Luki
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: Database unavailable
I believe this refers to the AstDB not the mySQL database. Make sure
the astdb file is writeable by user asterisk. The file is usually in
/var/lib/astdb.
--Luki
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delay before Answer fixes it. I tried tracking this down but didn't
have much luck. It's fine on outgoing calls.
--Luki
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. Like:
[provider-reinvite]
type=peer
host=external_sip_server.com
canreinvite=yes
...
[provider-noreinvite]
trype=peer
host=external_sip_server.com
canreinvite=no
...
exten = _1[0-4]X0.,1,Dial(SIP/{EXTEN:[EMAIL PROTECTED])
exten = _1[5-9]X0.,1,Dial(SIP/{EXTEN:[EMAIL PROTECTED])
--Luki
and
give yourself some feedback -- succeeded, failed because X / Y / Z.
Hope that helps...
--Luki
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probably will have to make your hands dirty and analyze where
ast_rtp_bridge() in rtp.c bails out. But since you are on a LAN it
makes the situation a lot easier.
--Luki
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Last time I checked, Broadvoice sent the Alert-Info header in the
INVITE message. The main line does not have this header, an add-on
line does.
On 1/22/06, Robert Mann [EMAIL PROTECTED] wrote:
Can * detect distinctive ringing on a SIP line? The reason I ask is I have
broadvoice with an add on
in app_rxfax that may shed some light. Otherwise see
Alexander's reply about the connection quality.
--Luki
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