> The config sorcery wizard is implemented by the res_sorcery_config.so module
Yup, that fixed it, modules.conf now starts with
[modules]
autoload=no
load => res_sorcery_config.so
load => res_pjproject.so
load => res_rtp_asterisk.so
;
Thanks!
--
And
ed or why I need it. Googling for this phrase suggested I
created an empty config file for pjproject but this also didn't
resolve this problem.
I am sure I must have missed something, can someone point me in the
correct direction?
Thanks!
--
Andreas Sikkema
--
> but as soon as I configure another sip registration on another server,
> outgoing
> calls drop after 32 seconds.
Are both your servers behind the same NAT router?
--
Andreas Sikkema
--
_
-- Bandwidth and C
t source port for SIP traffic. This should be applicable
to most UDP based protocols.
I think this is valid for most routers below a certain price point
($250?), perhaps those running Linux might not be affected.
--
Andreas Sikkema
--
_
tions.
I've never played much with the Queue command so I don't know if there's
a flag that is needed to add ringing from the Queue command or that a
simple additional Ringing command in latina_open might help.
--
Andreas Sikkema
--
_
t certified because
Open Source", was basically their answer.
--
Andreas Sikkema
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar eve
PSTN
> converter.
It's actually pretty easy.
If an INVITE message has a tag parameter in both To and From headers,
it's a re-INVITE. If the To header doesn't have a tag parameter, it's an
initial INVITE.
--
Andreas Sikkema
--
__
I don't have access to to do a quick test :-(
--
Andreas Sikkema
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these implementations would have worked fine when the called
party would have just ignored the offending media stream, instead of
sending an explicit deny.
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Andreas Sikkema
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Hi,
What are the recommended T.38 settings for sending/receiving faxes
from Cisco AS5350XM gateways? The chan_sip.conf file has a remark
about what Cisco is doing wrong and says that the values received from
the gateway should be overridden, but doesn't say what settings to use
for maximum success
g using an ACS..
--
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asterisk
> That's my question...the sbc provides security over trunking, right? The
> same can do Asterisk or a Proxy..isn't? Does an SBC can provide any kind of
> add-value to an Asterisk deployment?
A PBX provides functionality to users. An SBC *can* secure a PBX
against the outside world, but that is co
ay around is not that different.
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d up to
6 repeaters around one basestation.
Extending your range beyond that requires a proper DECT network and
brings you into a whole new cost level. But that can go up to 256x12
handsets and 256x8 (IIRC) simultaneous calls...
--
Andrea
> deny=0.0.0.0/0.0.0.0
> permit=XXX.XXX.X.X/29
> permit=192.168.1.0/24
Are you sure your provider *always* sends data from this /29?
Maybe you have this in your iptables as well and sometimes audio is
received from outside this /29 and therefore blocked?
--
Andrea
On 1/13/12 2:32 PM, Jonas Kellens wrote:
> So the context TrunkAccounts is not included.
>
> Do you know why ?
Does reloading the dialplan (dialplan reload) give any useful output
relating to these two contexts?
--
Andrea
> [root@haddock8-astrx dahdi-linux-complete-2.5.0.2+2.5.0.2]# make all
> make -C linux all
> make[1]: Entering directory
> `/usr/src/dahdi-linux-complete-2.5.0.2+2.5.0.2/linux'
> make -C drivers/dahdi/firmware firmware-loaders
> make[2]: Entering directory
> `/usr/src/dahdi-linux-complete-2.5.0.2+2
o there's no
real need to redefine it in the SDP, especially not since the answering
party already knows that the initiating party also uses the same value.
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ages but I just
doesn't seem to be able to make it work.
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s calls. If you add it all up we were actually
buying more DSP channels than E1 channels were available, for some
reason Cisco designed the machine like this, perhaps to cover for slow
call teardowns occupying DSPs too long.
--
Andreas Sikkema
--
___
e way to be absolutely sure
what Asterisk is trying to do. Everything else is just guessing.
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On 6/20/11 5:19 PM, Lyle Giese wrote:
> That's why other free providers don't use SIP phones, but build
> their own client software.
Real SIP providers fix this for their customers.
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Andreas Sikkema
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_
eem to have is probably related to them using the
same software for a simple 2xFXO port gateway as those for 4xISDN BRI or
4x ISDN PRI.
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On 5/4/11 7:10 PM, John Hablitzel wrote:
> exten => xxx,n,Set(CALLERID(name)=)
I'd either leave the name alone or do te following (haven't had the need
for removing it):
exten => xxx,n,Set(CALLERID(name)="
seems to be the answer to this, but I can't seem to get it to work
> right. Any ideas?
It's been years since I used GNUGk, but I'd check the mailinglist at
http://www.gnugk.org/ The core developers have always been very he
On 4/28/11 5:25 PM, Bruce B wrote:
> Is there any easy way to simulate a distorted SIP line temporarily for
> testing?
Build a Linux based router and use netem/tc to mess around with the
routed traffic. You can insert packetloss, jitter, etc and have it be
reproducable.
--
Andreas S
d then there's the apartments, it could get *very* expensive when you
need to replace wiring and the "phones" in each apartment to something
VoIP like.
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ike
> this :
> /Contact: /
Change your register line into this:
register => 33:mypass@ip_sip_server/33
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New t
, but when I tested
http://robin.nl/en/products/robin-compact-sip/ it worked flawlessly; I
don't have a doubt it will work with Asterisk.
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Muting the microphone on the Android side did not solve the really loud
second echo which suggests to me there might be something of a loop in
the operating system looping the audio back.
The phone is a Sony Ericsson Experia model, no idea what the exact type
th endpoints, is not
unheard of.
Network bandwidth is not a very good indicator of the quality of your
network Make sure you know if there's packet loss on individual links
(managed switches FTW), what the jitter is end to end, etc.
--
Andreas Sikkema
--
ots of statistics, including packet loss, jitter, etc. Check the
Wireshark site (http://www.wireshark.org/) for more information.
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New to As
ps this is a result of
some sort of SIP ALG or a stupid basic NAT implementation to reduce code
complexity on the router, but it is a nuisance either way.
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Andreas Sikkema
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onsible (development, maintenance, support) for an
Asterisk based VoIP platform providing a replacement for
residential PSTN lines. So I'm technically just a user ;-)
I've literally got _thousands_ of users and Asterisk is rock
solid for us.
--
Andreas Sikkema
___
rver version for the right syntax
> to use near ') FROM tblCall WHERE id=7' at line 1
I think the solution would be to escape the , with a backslash, so
your query would look like this:
SELECT TIMEDIFF(callend\,callstart) FROM tb
same username is, IIRC, more or less default
behaviour.
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but they are all doing very I/O intensive stuff.
I frequently see machines doing loads of over 4 with total CPU
load not above 100% (of 400% possible)
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asterisk-us
would be my way.
Using MYSQL() (or equivalent) heavily in the dialingplan is IMHO the
nicest way of doing things like this. You can do lots of simultaneous
calls before getting into trouble.
Appending stuff to the CDR userfield is just plain ugly and asking for
trouble (are you sure you c
is just as you
> expect it to be.
Well, they're called macro's for a reason You guys are
proposing adding functions or procedures.
My first step in any macro would be to copy incoming
variables, be it arguments or even asterisk defined stuff
to local variables. But that is j
is still active" CDR
- and maybe start/stop (a "continuation", see above) cdrs, instead of
just one...
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> However, even once I reloaded the extensions, its still only
> using ulaw.
You didn't reload the sip config? Maybe that's your problem?
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asterisk-users
> I also dropped the quotes on the dnis=${IVR-Exten}.
That's only allowed if the dnis column doesn't contain a string.
--
Andreas SikkemaBBeyond
Software EngineerPlaneetbaan 4
+31 (0)23 70743422132
l 1850's and I couldn't be happier
with them. But then I don't use any Digium, Sangoma or other
cards. We're running 100% VoIP through them.
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asteri
If you have no statuc stuff in your dialplan, how do you use the 'include =>'
statement? We don't have users... we have companies. It's a hosted IPT
service... and to make the problem even more insane, each company has multiple
levels of organisational structure.
Hardly, you're not required to
have no
static stuff in our dialingplan. And we have quite a number of
users.
But no queues etc.
--
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___
--Bandw
ing destination)
This will not screw up your extesnions matching, but you will
need to check that outgoing_callerid has been filled before setting
callerid (or make sure it is always filled with something sensible).
Check the variables page in the wiki on exact syntax ;-)
--
Andreas Sikkema
her. For some reason this involves stopping the
existing audio, waiting a little while and then starting a new
audio stream.
So far this one of the reasons why I don't like reinvite...
--
Andreas SikkemaBBeyond
Software EngineerPlaneetbaan 4
+31 (0)2
as apache, so I
start it around the same time using the same priority as apache and as
far as I know networking should work at that time or not at all, not
somewhere in between.
pebkac?
--
Andreas SikkemaBBeyond
Software EngineerPlaneetbaan 4
+31 (0)2
caller ID strings? Is it expected to be able to
> also handle E164
> numbers (which can be up to 15 digits) as well, or is there
> another method for
> that?
Sure, no problem. As another reply said, it's just a number.
--
Andreas SikkemaBBeyond
Software Engine
other firmware, because I just
can't get it registered at all, let alone make calls.
We do have proxies for RTP ;-)
--
Andreas SikkemaBBeyond
Software EngineerPlaneetbaan 4
+31 (0)23 7074342
to solve this using STUN etc,
while I would prefer they also wouldn't have the software
care if it is on the inside of a NAT like most other CPE's
so our platform can take care of things.
--
Andreas SikkemaBBeyond
Software EngineerPlaneetbaan 4
+3
be more precise).
On H.323 this is called "early media" and in SIP this would be
signalled using a 186 Session Progress message.
--
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___
> Andreas Sikkema wrote:
> >
> > Hi,
> >
> > To combine two sources of CDR's I want Asterisk to save the
> SIP callid for
> > all calls. I know there's a variable that contains the SIP
> CallID value,
> > but is this the callid va
y the same? (I've not yet checked a trace, I'm sorry for
that). I've tried to read chan_sip, but couldn't find something in the time
I had today. I've found hardly any documentation o this variable, apart from
that it exists and that it contains "the" SIP CallID
> I believe what you refer to is called "Ring Back When Free"
> at least thats how I know it in the UK.
Ah yes, no I remember. We called it "Automatic Ring Back".
So we had "normal ARB", or "ARB on next use".
--
Andreas Sikkema
no answer
The second one is tricky; after the destination number has
been used again, the switch will dial the originator and
then the destination and connect the two legs.
--
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Software EngineerPlaneetbaan 4
+31 (0)23 7074342
orks reasonably well, it seems to detect 99% of the major
problems.
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___
--Bandwidth and Colocation provided b
update list of packets in real time" and
"automatic scrolling in live capture"
A "sip" display filter is needed so you only see SIP traffic,
a sip capture filter might be needed for very busy networks
--
Andreas Sikkema BBned NV
Software Engineer
these options should be
available whenever Asterisk generates _any_ media by
itself, including conferencing.
IVR functionality and the like become much better when
ztdummy or another timing source supported by Asterisk is
available.
--
Andreas Sikkema B
IP "ATA" is configured to
use rfc2833 but is also a little to lote with the filtering
out of the DTMF. So sometimes it's not Asterisks fault at
all ;-)
And then there's some IVR's that don't notice it at all, while others
are totally unusable.
--
Andreas Sikkema
a longer time is assigned to a new and
appropriately named variable. So the original variable can be used
again.
We've got loads of queries in our extensions.conf.
--
Andreas Sikkema BBned NV
Software EngineerPlaneetbaan 4
+31
s, but that is not very
reproducable, dependencies listed in the rpm file (or equivalent)
usually takes care of this. When isntalling from source, you're on your
own.
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Software EngineerPlaneetbaan 4
+31 (0)23 7074342
> There should be other "voices" worth while...
>
> Give other people the chance
>
> The market is growing...
>
> Be open :)
I'd _love_ a different voice for the default
distribution. To my (European) ears Allison
is practically in
as verified.
Do you have the Digium G.729 codec installed? This one provides "show
g729
"
I have no idea if the IPP hack provides a similar interface.
--
Andreas Sikkema BBned NV
Software EngineerPlaneetb
here's an error in your configuration somewhere.
Host names cannot contain , characters.
--
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Van Vollenhovenstraat 33016 BE Rotterdam
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--Bandwidth and Colocati
Part3 = ${myNumber:7:3}
exten => s,4,SetVar(myNumber = $strPart1$strPart2$strPart3
But I'm using quite an old Asterisk, so current syntax might
be a little different, but the Wiki suggests this still works.
--
Andreas Sikkema bbned NV
Van Vollenhovenstraat
exact same SIP message types to you
as provider2? It looks to me like provider1 is not sending
a 183 Session Progress message. Which is usually used for
this kind of functionality I think.
--
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Van Vollenhovenstraat 33016 BE Rotterdam
t: +31 (0)10
have
hardly any outside consequences...
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Van Vollenhovenstraat 33016 BE Rotterdam
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. The person calling from PSTN is not getting any echo.
Make sure you're not playing the recorded sound from your
microphone back to your loudspeakers.
--
Andreas Sikkema bbned NV
Van Vollenhovenstraat 33016 BE Rotterdam
t: +31 (0
Sherwood McGowan wrote:
> Has anyone else had problems with users being able to press key
> tones during a voice prompt? I have a few users complaining that
> some systems will not recognize key presses during them.
You are using Backgroudn() to play the prompts?
--
Andrea
[EMAIL PROTECTED] wrote:
> What is CFU and CFNR?
Call forwarding unconditional
call forward not reachable
--
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Van Vollenhovenstraat 33016 BE Rotterdam
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rd you don't need it.
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Asteri
rea where lots of people with
dect handsets gather (meeting rooms, canteens) you
need lots of basestations.
I've worked in a building where there seemed to be an
overkill of basestations every hallway had 3 or 4, (every
20 meters or so) and still there were areas with
insufficien
use the voicemail function to record, the
> message is not recorded (0 byte file is created) and it gives the
> following errors -
>
> "unable to convert from g729 to slin"
You can force Record to record to G.729, but I'm not sure the
voicemail application h
Chad Brown wrote:
> I'm publishing tftp through my firewall to support external Cisco
> 7960 sip phones.
I hope the files requested by the Cisco phones don't contain username
/ password information. Passing that in cleartext is just so wrong ;-)
--
Andreas Sikkema
Hi,
Has anyone written a SPEC file for zaptel, with kernel
2.6 and udev support? I can find some spec files here
and there, but from what I can see they're all kernel
2.4 / non udev...
--
Andreas Sikkema bbned NV
Van Vollenhovenstraat 33016 BE Rotterdam
t: +31
[EMAIL PROTECTED] wrote:
> And it's a great shame Digium hardware has such problems on
> Dell kit, since
> there's so much of it about :(
If you don't use digium hardware, there's of course no problems with using Dell.
--
Andreas Sikkema bbn
es?) providing access
to a database which telcos can use to find the rates on this kind
of numbers.
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Van Vollenhovenstraat 33016 BE Rotterdam
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up. We
had no control over the distro installed on the blade.
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) from the
blades, but not because blades don't work. We really liked them.
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Aste
, if it is not from
asterisk, it must be meant to go to asterisk.
Add a couple of other tests (known user, etc) to it and then I
think you'll have what you're looking for.
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t: +31 (0)10 2245544
only modprobe zaptel first would help.
(Debian sarge)
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ht
n. Call it something
else (iptel-out?) and maybe that solves your problem.
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oaded or anything.
I see these with every single call. I (naturally I'd say) also
have reports of dropped calls, but have never been able to relate
them to these messages. The messages happen much more often.
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S0]-> Asterisk <-[SIP]->
> Final users.
>
> Is it possible?
> does Asterisk support QSIG and S0 interfaces?
As far as I know, Asterisk doesn't support QSIG. Do you
_have to_ use QSIG?
I'd just use a PRI interface (DTU-PH IIRC) to connect to
Aster
tches that it supports.
On the (admittedly relatively old version we're using) this will
only work when I'm logging in via SSH. When working from the
console -n doesn't work.
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Andreas SikkemaRits tele.com
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t: +31 (
I'm using. I'm not using this card for
anything at all, but I'm wondering how to set it up for
timing only. What do I have to do (I have no experience
at all with zap channels and the zaptel.conf file)?
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Van Vollenhovenstraat 3
om a third party.
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hine to only allow SIP traffic from
the machine with SER?
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Hi,
What does this line of output mean?
Bridge stops because we're zombie or need a soft hangup:
I'm seeing this sometimes... I've looked in channel.c,
but the code is not much more revealing than the
debug line...
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Andreas SikkemaRits tele.com
Van Voll
s a link to the google cache copy:
http://66.102.9.104/search?q=cache:pR1IMCaiRcQJ:www.ramdyne.nl/index.php%3Fcat%3D11+%2Basterisk+%2Bramdyne+%2Bdebian&hl=nl&start=1
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[EMAIL PROTECTED] wrote:
> Is there anyone else with the same problem?
Yes, we've seen the same problem. We have found a work
around, but I'm unable to to look into it today.
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t: +31
s erratic. I've
got to solve some other problems first, but Asterisk T.38 pass
through is the next major issue.
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__
ng on Fax as well and if I'm not mistaken, there is a mode
where Asterisk doesn't have to know very much about T.38 to make it
work.
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t: +31
[EMAIL PROTECTED] wrote:
> (side note: If you havent bought their hardware and are using
> Asterisk for "free" them again you should expect even less
> assistance imo)
Right, so I have to buy hardware I don't need?
--
Andreas SikkemaRits tele.com
Is this normal? How do I prevent neeeding
this step?
I know we shoul move to at least 1.0, but we're
running this in production and we haven't felt the
need to upgrade. If necessary I can backport...
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Andreas SikkemaRits tele.com
Van Vollenhovenstraat 33016
I'm wondering if they didn't mean a "pseudo" ani?
Are you sending internal Asterisk ANI or the ANI
Gobal Crossing is expecting?
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t:
[EMAIL PROTECTED] wrote:
> The invite message is sent as a single message to asterisk
> containing the whole number string, as apposed to each number
> individually.
Does SIP support non en-bloc dialling mode?
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Van Vollenhovenstraat 3
gt; circuit, then it has only 23 channels to carry voice, as the 24th
> channel is used for the D-channel (signalling channel).
Only if you're in the US. We have 30 + 1 :-)
--
Andreas SikkemaRits tele.com
Van Vollenhovenstraat 33016 BE Rotte
e? :-))
A combo of SER and Asterisk is pretty powerfull IMHO.
--
Andreas SikkemaRits tele.com
Van Vollenhovenstraat 33016 BE Rotterdam
t: +31 (0)10 2245544f: +31 (0)10 2245540
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