> 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
sted 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!
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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?
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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.
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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.
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, was basically their answer.
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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.
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fine when the called
party would have just ignored the offending media stream, instead of
sending an explicit deny.
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New to Asterisk
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
..
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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
is not that different.
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asterisk
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...
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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?
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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?
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[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
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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New
to make it work.
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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.
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is trying to do. Everything else is just guessing.
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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)=)
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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.
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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 helpful
to me.
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you
need to replace wiring and the phones in each apartment to something
VoIP like.
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Change your register line into this:
register = 33:mypass@ip_sip_server/33
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://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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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.
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, including packet loss, jitter, etc. Check the
Wireshark site (http://www.wireshark.org/) for more information.
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or a stupid basic NAT implementation to reduce code
complexity on the router, but it is a nuisance either way.
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) 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.
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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 tblCall WHERE id=7
Maybe even the brackets ()
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frequently see machines doing loads of over 4 with total CPU
load not above 100% (of 400% possible)
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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 can always separate the different values?).
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, 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 just me and my coding
convention.
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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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I also dropped the quotes on the dnis=${IVR-Exten}.
That's only allowed if the dnis column doesn't contain a string.
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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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before setting
callerid (or make sure it is always filled with something sensible).
Check the variables page in the wiki on exact syntax ;-)
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dialingplan. And we have quite a number of
users.
But no queues etc.
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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
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...
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, 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?
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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.
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,
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.
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just
can't get it registered at all, let alone make calls.
We do have proxies for RTP ;-)
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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 value of the incoming INVITE message or the
outgoing
message
? (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 value.
Can anyone enlighten me?
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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.
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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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99% of the major
problems.
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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.
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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
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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.
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named variable. So the original variable can be used
again.
We've got loads of queries in our extensions.conf.
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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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? This one provides show
g729
I have no idea if the IPP hack provides a similar interface.
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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 incomprehensible.
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somewhere.
Host names cannot contain , characters.
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= ${myNumber:4:3}
exten = s,3,SetVar(strPart3 = ${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.
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Van
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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is not getting any echo.
Make sure you're not playing the recorded sound from your
microphone back to your loudspeakers.
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consequences...
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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?
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[EMAIL PROTECTED] wrote:
What is CFU and CFNR?
Call forwarding unconditional
call forward not reachable
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[EMAIL PROTECTED] wrote:
I just setup a Dell 1800, not a 2850, which is working
awesome.. only had to
disable USB, which realistically no-one on a phone system
would care about
anyways.
Oh, really? Only if you're running a 2.6 kernel or using
a zaptel card you don't need it.
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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
insufficient coverage...
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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 has the same possibility.
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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 ;-)
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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...
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t: +31 (0)10
[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.
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access
to a database which telcos can use to find the rates on this kind
of numbers.
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blades don't work. We really liked them.
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wiki)
If you have no zaptel hardware and must rely on software you should
use kernel 2.6's ztdummy, don't you? It is better, and also does
not rely on USB.
Yes, true, but this entirely depends on how the blade is set up. We
had no control over the distro installed on the blade.
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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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)
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?) and maybe that solves your problem.
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(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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.
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
Asterisk with a sutable PCI card in the server.
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) this will
only work when I'm logging in via SSH. When working from the
console -n doesn't work.
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. 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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t: +31 (0
SIP traffic from
the machine with SER?
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.
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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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[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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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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some other problems first, but Asterisk T.38 pass
through is the next major issue.
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?
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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[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?
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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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[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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Andreas SikkemaRits tele.com
Van Vollenhovenstraat 33016
voice, as the 24th
channel is used for the D-channel (signalling channel).
Only if you're in the US. We have 30 + 1 :-)
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Andreas SikkemaRits tele.com
Van Vollenhovenstraat 33016 BE Rotterdam
t: +31 (0)10 2245544f: +31 (0)10 2245540
lots of this kind of problems before. We
had lots of stability problems with GNUgk on Debian
Woody. Once we moved to Sarge we had no problems at all,
with uptime going from a couple of days to several
months when we had no need for GNUgk anymore.
--
Andreas SikkemaRits
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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Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo
[EMAIL PROTECTED] wrote:
In some SIP invite messages I see the below codec negotiation
string, I am wandering what the 101 telephone-event/8000
means Which codec is that?
RFC 2833 DTMF events
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Andreas SikkemaRits tele.com
Scheepmakersstraat 11 3011 VH Rotterdam
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