I've seen that behaviour on he MV-374. One possible solution (workaround)
is to prevent the gateway from registering itself and to declare each of
the channels explicitly in sip.conf via its associated IP + port.
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On Wed, Nov 25, 2009 at 11:41 PM, Shaun Clark shaun_cl...@hotmail.com wrote:
Can you add an agent dynamically to a queue with an external number, i.e.
cell phone as an extension? If so how? Thanks!
Maybe adding the channel Local/PSTN-number@context-that-dials-PSTN
to the queue as a member
Oliver,
Without any experience with Asterisk 1.6.1.x here it goes:
- Is your signalling=pri_net correct or should it be pri_cpe ?
- I'm having myself some issues with DAHDI 2.2.0.2 + TE121
which, I reckon, is equal to your TE122 apart from the
bus interface (PCIe vs PCI) -- see the
Shaun,
Thanks for your feedback. See my inline comments.
On Fri, Nov 13, 2009 at 7:18 PM, Shaun Ruffell sruff...@digium.com wrote:
It appears there may be a regression in dahdi-linux 2.2.0 with regards to
the wcte12xp driver and the VPMADT032 module (as discussed
Although I've never tested such feature on those devices, I know
that it was only enabled in a recent firmware (6.1.3a/6.1.5a ?).
Are you running it ?
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Hi Asterisk Users,
We've been experiencing some tough time regarding a new Asterisk installation
connected to the PSTN via an ISDN PRI with a Digium TE121 with the optional
VPMADT032 echo cancellation module.
For now, I'll focus on something very specific which is summarized on this
2009/9/27 Leif Neland le...@neland.dk:
Can I, via a callfile, or command-line parameters to Asterisk start a
dialplan-script?
eg asterisk -someflag execute callalert
then in dialplan
[callalert]
exten = s,1,SIPAddHeader(Call-Info:\;answer-after=0)
exten = s,2,dial(SIP/36)
exten =
On Sun, Sep 27, 2009 at 12:07 AM, Gabriel Ortiz Lour
ortiz.ad...@gmail.com wrote:
Hi,
Is there a way to know for how long an agent is talking on the queue call?
(without keeping a timer myself... just asking asterisk)
Identify the channel at the CLI and then get its details via
core
I've been willing to give such a solution a try but the lack of time has
prevented it to date...
Are you using realtime for your SIP peers/users ? Would the failover
behaviour improve under such scenario ? (just a thought)
--
exvito
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The system specs mention PCIe expansion slots, so your only
option is the TE420B.
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On Fri, Apr 3, 2009 at 11:11 AM, Richard Brady rnbr...@gmail.com wrote:
Exvito
Did you ever make any progress on this?
...no, sorry. Never got to the perfect solution. (and in all due
honesty, I can't
recall the exact setup we ended up deploying)
--
exvito
# rasterisk
Connected to Asterisk 1.4.23 currently running on debian (pid = 17191)
samuel*CLI restart now
samuel*CLI
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk ending (0).
# rasterisk
Connected to Asterisk SVN-branch-1.4-r178373 currently running on debian
On Tue, Feb 17, 2009 at 8:04 PM, Ray Chen ray1...@techie.com wrote:
I have a problem of using call file to make an auto dial out call through
FXO channel. I defined the channel in the call file as Channel:
DAHDI/1/8775203463 When I put the call file under the
/var/spool/asterisk/outgoing dir
On Thu, Feb 5, 2009 at 7:22 AM, Geoff Lane ge...@gjctech.co.uk wrote:
The nice thing about that is that if I use MySQL I can run the
management application on another machine, and so don't need to run a
web server on the Asterisk box. However, I wonder whether the overhead
necessary to run
App nvfaxdetect() works fine for that purpose on both Zap and mISDN.
See http://www.voip-info.org/wiki-NVFaxDetect
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On Tue, Feb 3, 2009 at 5:04 PM, Vieri rentor...@yahoo.com wrote:
I did but apparently, there's nothing in the guides that lets me do this.
It's something about supporting 484 responses that Grandstream GXW4008
seems to do and Linksys SPA8000 doesn't (or at least it's not documented).
In
On Wed, Feb 4, 2009 at 7:40 PM, Jerry Geis ge...@pagestation.com wrote:
Seems like the first call to Channel is being MADE successfully.
Then it goes to do Context and Exten: I get failed...
[smvoice-dialout]
exten = smvoice_single_mediaport,1,agi(smvoice)
exten =
For a simple (but flexible) case I would consider ODBC + func_odbc.
Here is the idea (in case you aren't aware of how it goes...)
- Make a DB available (your choice as long as it is accessible via ODBC)
- Create table in it with your contacts (say columns number and
name, maybe more)
-
Any suggestions?
Jerry
Are you sure asterisk is to behave as signalling=pri_cpe or should it
be pri_net ?
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On Mon, Feb 2, 2009 at 8:39 AM, Idris AVCI idris.a...@vodatech.com.tr wrote:
In my situation AMI is not an option. When somebdy puts a call on hold, on
asterisk console I can see messages like Started music on hold, class
'default', on SIP/ and Started music on hold, class 'default',
(my 2c, Portugal Based)
- Most really small installations are PtMP (that's the default you
get when ordering a BRI)
- You also get 3 MSNs and an NT.
- You can order a TA instead of the NT.
- You can order PTP + optional DDIs in blocks of 10, but you need to
be explicit.
- Larger
On Tue, Jan 27, 2009 at 10:21 PM, Pascal Bruno tipas...@gmail.com wrote:
Is it possible to retrieve the DIALSTATUS variable when placing call through
a call file. This variable is set when using the Dial() application from
the dialplan, but I am using a call file for my current application and
On Fri, Jan 30, 2009 at 3:23 PM, Danny Nicholas da...@debsinc.com wrote:
The dialplan AFAIK doesn't cover HOLD handling. If you can spare the
overhead, you can make a daemon to watch hints and run a script whenever the
hint for a line goes to hold and changes from hold to inuse. Just run
On Thu, Jan 29, 2009 at 6:15 PM, Vieri rentor...@yahoo.com wrote:
I'm trying to do the same in the SPA8000 units but without any luck. If
anyone is doing something similar with this device then I'd appreciate it if
you could share your relevant config options (dial pattern, etc.).
Not
Assuming you are using SIP phones and IIRC, you can hint at the
codec to be used by setting the SIP_CODEC variable in the dialplan;
before Dial()'ing, of course ! :-)
I think this is still an area where asterisk needs improvement... Dynamic
codec (re) negotiation. Anyone care to correct
Anyone done anything similar before that would care to share ?
Here is a snippet of our standard ks file for automated installs:
%packages --nobase
@core
sendmail
sendmail-cf
ntp
vixie-cron
crontabs
at
logrotate
telnet
bind-utils
lsof
wget
which
unzip
man
bc
nc
sharutils
# Up till now, just
While I don't know the OpenVOX B200P specifics, some interface cards
need you to change physical jumpers in order to acheive NT vs TE, mode.
Could that be the case ?
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On Mon, Jan 5, 2009 at 8:20 AM, Nick Wolf new...@gmail.com wrote:
besides this, I paste my zaptel.conf :
span=1,1,6,ccs,hdb3
span=2,1,6,ccs,hdb3
bchan=1-15,17-31
dchan=16
bchan=32-46,48-62
dchan=47
loadzone=fr
defaultzone=fr
when I put 6 in LINE BUILD OUT (1,1,6,ccs,hdb3) value I got
On Mon, Jan 5, 2009 at 10:12 PM, David fire ddf...@gmail.com wrote:
if you don't know any parser maybe you can send me a link or a pdf whit info
on how to parse the log.
...check queuelog.txt under the doc/ directory on the asterisk source
distribution (apparently, under 1.6 it is
On Sun, Jan 4, 2009 at 8:37 AM, Nick Wolf new...@gmail.com wrote:
I forgot to describe the audio problem, well, I experience micro cuts in the
voice, this does not happen during the whole call, it happens during 2
seconds then audio becomes normal, then back again 2 or 3 seconds then
goes
On Sat, Dec 20, 2008 at 5:54 PM, Jerome Deyle jde...@gmail.com wrote:
Running AsteriskNow, with FreePBX front end.
Have two users who use softphones on notebooks in the field. Problem is that
if the softphone receives a call, but the user is not available to pick it
up, Asterisk will send the
On Tue, Nov 4, 2008 at 3:12 PM, Igor Zamocky [EMAIL PROTECTED] wrote:
http://bugs.digium.com/view.php?id=13645
Thanks Igor, we'll keep an eye on it.
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Hi list,
I just experienced an odd behaviour in 1.4.22 vs 1.4.21.2.
To cut a long story short, IAX2 is not tx-ing hangup...
Scenario is composed of two asterisk systems A and B.
A receives calls from IAX users X, Y, Z, etc, does some
validation and forwards them to B, also over IAX.
When B
...as others have mentioned, yes they do have the ability to be centrally
managed, provisioned, configured.
Also, from the latest firmware, 6.x.x:
- Ability to use line buttons as quick dials
- Ability to query centralized LDAP for directory (I haven't tested
this one yet)
So, the
So let's say, you've got :
a perfectly running tftp server somewhere on your LAN,
it holds foo.txt file in its /srv/tftp directory.
Which command could you type in for a LAN workstation to receive this
foo.txt ?
tftp is the client, do you have it installed ?... example:
# tftp
I need to have different gain settings on each channel. Is this easy to
achieve?
txgain, rxgain and many other parameters are defined on a per-channel
basis in zapata.conf, they're not global. Each channel definition
channel = x
assumes previous definitions of such parameters.
I recently observed a similar behaviour under 1.4.21. The member
was a SIP phone which had its calls forwarded to another SIP
phone via its built-in configuration... (fyi: linksys spa922)
For some reason, asterisk could not manage this scenario. I still
have to test it better to
Here is an update,
1. Reviewed 'core show locks' with the help of russellb @ #asterisk-devs
last friday
2. Recommended recompilling asterisk with DONT_OPTIMIZE and
getting a stack trace with:
# gdb /usr/sbin/asterisk $(pidof asterisk)
(gdb) set pagination off
On Wed, Jun 11, 2008 at 12:33 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
Try switching from IAX to SIP.
Steve, thanks for your suggestion... As you may understand that is not an easy
decision to take and implement: we're peering with about 20 other systems
within a private network where
On Fri, Jun 6, 2008 at 1:01 PM, Ex Vito [EMAIL PROTECTED] wrote:
In Our Heads
--
- we're suspecting that the presence of the TC400B is making asterisk behave
in different ways that lead to what we're now calling a hang (that is the
apparent change in the system since
On Fri, Jun 6, 2008 at 3:16 PM, Shaun Ruffell [EMAIL PROTECTED] wrote:
I'm soon going to petition for this interface to be merged into the trunk, so
if you would like to try the branches out now and need any help, please
contact me directly.
Thanks for you feedback Shaun.
I've had a
On Fri, Jun 6, 2008 at 5:01 PM, Andres [EMAIL PROTECTED] wrote:
Of course, future possibilities of changing codecs, removing the TC400B
or others are open (such as: I guess we enabled the 1st voicemail
account as test
on the same day that we installed the TC400B -- could it be the change
On Tue, May 20, 2008 at 2:14 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Tue, May 20, 2008 at 8:55 AM, Cavanna, Richard [EMAIL PROTECTED] wrote:
...
Cards I have installed:
Digium TE205P - 5v
TDM410
I hear rave reviews about Supermicro but no personal experience.
I like
On Thu, May 8, 2008 at 3:59 AM, Russell Bryant [EMAIL PROTECTED] wrote:
Ex Vito wrote:
Now, how to move on to acheive some kind of fault tolerance ?
According to the docs we've studied, DUNDi does not like loops
(which we assume one can limit with low enough TTLs).
Which documentation
On Thu, May 8, 2008 at 1:23 AM, Benoit Plessis [EMAIL PROTECTED] wrote:
Tilghman Lesher a écrit :
Your question leads to this question: why don't you create a proxy
application that listens on AMI and populates a database outside of
Asterisk,
then do all your queries to that database?
Hi list,
I'm planning a private DUNDi network for a cross-country
distributed PBX. Initially it will be composed of about 10
systems, growing to about 20.
Current requirements point to a topology of two interconnected
DUNDi hubs, each peering with half the PBXs... This would
lead
On Fri, Apr 18, 2008 at 10:12 PM, Matthew Fredrickson
[EMAIL PROTECTED] wrote:
Ex Vito wrote:
Matthew,
...is there any specific test you'd like us to perform on this revision ?
(considering that currently we have no PSTN line to attach to... we
can cross-connect
On Mon, Apr 21, 2008 at 4:38 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Ex Vito wrote:
...when can one expect to have your new code available in a zaptel
release ?
In the next one or maybe later because the branch you're working on
has lots of different things
We've been very happy with the SRW224Ps we've deployed.
(noisy as hell... good for either the datacentre / computer room or
for installation in a noise-cancelling cabinet... but then again, are
there any PoE switches that aren't ?)
--
exvito
On Fri, Apr 18, 2008 at 4:15 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
I just realized where this is coming from. I was attempting to patch
this from a different angle, but as soon as you mentioned the drastic
difference in load time I realized what had happened. I'm going to make
On Fri, Apr 18, 2008 at 8:20 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
I just updated the branch. Wait about 5-10 minutes in case for the
changes to get mirrored, and then try updating and doing it again.
Looks better, no more soft lockup and ztcfg time is comparable to
On Fri, Apr 18, 2008 at 9:36 PM, Ex Vito [EMAIL PROTECTED] wrote:
On Fri, Apr 18, 2008 at 8:20 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
I just updated the branch. Wait about 5-10 minutes in case for the
changes to get mirrored, and then try updating and doing it again
On Wed, Apr 16, 2008 at 7:18 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Ex Vito wrote:
Tested with no 4K stack kernel and stackcleanup svn branch
zaptel version. Correct, the kernel no longer complains about
the soft hangup.
However the system still hangs (console
On Thu, Apr 17, 2008 at 2:36 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Apr 17, 2008 at 02:20:57PM +0100, Ex Vito wrote:
- Should this be considered a regression ?
Yes, it is a regression, and thus a bug.
Mattf has already offered you to work with him on resolving
On Tue, Apr 15, 2008 at 7:07 PM, Shaun Ruffell [EMAIL PROTECTED] wrote:
Your stack trace appears to possibly be stack corruption.
Could you try either this branch:
http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/
Just tried it... Behaviour looks equivalent. Drivers
Could be this...
http://www.misdn.org/index.php/FAQ_chan_mISDN#Why_does_the_L1_goes_DOWN_on_my_PMP_Isdn_Link.3F_Or_why_do_i_get_No_free_chan_even_after_group_call_from_chan_misdn_if_dialing_out_on_my_PMP_Link.3F
Hmmm... that's a long link. It is the
Why does the L1 goes DOWN on my
On Wed, Apr 16, 2008 at 3:26 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
The softlockup indicator should be benign. It gets called when loaded
the firmware for the part since the firmware image is so large and it
takes a long time to load. However, I might have a fix for you.
Can
On Wed, Apr 16, 2008 at 4:20 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
One thing also I would like to see is your kernel .config file. Another
thing that would for sure remove that warning is to disable the kernel
softlockup detector which is giving a false lockup warning in this
http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/
Question:
- The url you suggest is very similar, are we talking about
a different stackcleanup branch ?
Try:
http://svn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup/
Try the
On Wed, Apr 16, 2008 at 4:46 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
It's the same. Sorry, I sent you that email before I saw his message.
I just got an idea for a clever way to make the softlockup detector not
complain. I'll let you know when I have a patch to try.
...sure.
update with no 4K stack kernel:
- The kernel was build from stock centos 5 kernel 2.6.18-53.1.14.el5
- The only .config change was to disable the CONFIG_4KSTACKS
Tested zaptel-1.4.10, 1.4.9.2 and the stackcleanup svn branch as
suggested by Shaun and Mathew.
Short: Results are about
On Wed, Apr 16, 2008 at 6:51 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
One thing you can also do is pass the nosoftlockup kernel parameter
into the kernel from the bootloader. That should disable the softlockup
detector.
Tested with no 4K stack kernel and stackcleanup svn branch
Your stack trace appears to possibly be stack corruption.
Could you try either this branch:
http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/
Or with a kernel that does not have 4K stacks enabled? You can check if
your installed kernel does with the following
Or with a kernel that does not have 4K stacks enabled? You can check if
your installed kernel does with the following command.
$ cat /boot/config-`uname -r` | grep 4K
# CONFIG_4KSTACKS is not set
Opps, forgot to feedback: yes this kernel seems
to have CONFIG_4KSTACKS
On Tue, Apr 15, 2008 at 8:37 PM, Al Baker [EMAIL PROTECTED] wrote:
exvito - I know it is a pain in the cahoonkus - but would you consider
sharing the OTHER Digium board issues you are having , the recommended
steps you were given by Digium to troubleshoot them, and the results ?
I think
On Tue, Apr 15, 2008 at 11:37 PM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
In the sip peer definition,
disallow=all
allow=g729
allow=ulaw
SHOULD work. Asterisk can't transcode g729, so it should fall on ulaw
for the ZAP calls. But, when your polycoms talk with each
Hi list,
After a lot of testing + troubleshooting, I guess I'm observing
what I am now calling a regression with zaptel 1.4.10 (is it?)
As such I call for peer feedback, before either asking Digium
install support or filing a bug.
Thanks in advance!
System: HP Proliant DL380 G5
If you are running 1.4, check rtp-packetization.txt under
doc/ directory from source distribution.
Cheers,
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On Wed, Feb 13, 2008 at 6:49 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
Macros are deprecated. Gosubs are the way forward, and yes, they have
local variables. Simply define them once as Set(LOCAL(foo)=bar) and foo
will be gone when the innermost stack is removed (either by Return or
On Thu, Mar 13, 2008 at 3:47 AM, Gonzalo Servat [EMAIL PROTECTED] wrote:
I can't find any channel variable that gives me this info.
Gonzalo,
With SIP callers you can get the address from the SIPURI channel variable.
IAX does not seem to have an equivalent var... The best I could find is
Improvement: also check the funcions SIPCHANINFO and IAXPEER... With
this and the SIPURI channel variable you should be able to have all the info.
Cheers,
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On Thu, Mar 13, 2008 at 9:07 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
Which is something not unlike the Junghanns ISDNGuard.
...or the beroNet bero*fos
(https://shop.beronet.com/product_info.php/cPath/56/products_id/159)
Not affiliated, just a satisfied customer.
--
exvito
...as long as the destination does not answer you'll get
a NO ANSWER disposition.
Note, however, that answering can be one of:
- Dial a phone and the user answers the phone
- Connecting the caller to voicemail, for example,
after Dial timed out
- Playing an IVR / sound / music
Hi list,
I'm planning and testing a distributed asterisk deployment
throughout several sites; each will be connected to the PSTN
and all of them among themselves via IAX trunks. Phones
will be SIP.
I guess I already solved (worked-around, actually) asterisk's
codec negotiation
I don't have access to an asterisk system right now
(nor any other sort of information source) but I seem
to recall that from 1.4 onwards the config option for
recording queue calls is named differently...
Is it mixmonitor ? Check you 1.4 queues.conf sample.
PS: I'm not really sure
I wouldn't know how to do it the way you mention it,
via local channels...
Our implementation performs ringing transfers via
AMI redirect... The user action is performed on the
desktop, not on the ringing phone.
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On Tue, Mar 4, 2008 at 7:54 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Tue, Mar 04, 2008 at 03:05:43AM +, Ex Vito wrote:
...can you expand on that please ? I'm on my way to getting one of the
newer Digium TE220B PCIe dual T1/E1 to put on such a system. How
can I tell
On Tue, Feb 26, 2008 at 10:51 PM, Joshua Kinard [EMAIL PROTECTED] wrote:
Just don't use T1 cards w/ TigerJet chipsets in them on DL385's (and very
likely, 380's as well). I just learned this the hard way.
--J
...can you expand on that please ? I'm on my way to getting one of the
newer
I've struggled with this recently. In short:
- Observed behaviour is expected as of asterisk 1.2 and later,
as previously described by Mojo
- If you want to get the caller id for the channel calling (dialling)
into that channel for that specific Newstate: Ringing event, you
On Nov 26, 2007 4:06 PM, Steve Totaro [EMAIL PROTECTED]
wrote:
I wonder if anyone on the list has run a server with both types of cards
installed? Results?
Again, like Geert, not quite the same, but happily running quad BRI
(beronet, HFC based) + TDM400.
The BRI is running
explanation and I hope some one can shed some light into this.
Thanks a lot in advance for any insights! :)
Regards,
PS: I can provide simplified traces of the events if needed.
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On Nov 23, 2007 6:58 PM, Moises Silva [EMAIL PROTECTED] wrote:
I added the senddialevent, but not the condition you see below. That
one was added by someone else. It seems that determine wheter or not
the current extension will be set for outgoing calls. Setting
OPT_ORIGINAL_CLID may fix your
On Nov 20, 2007 6:13 PM, Philip Prindeville
[EMAIL PROTECTED] wrote:
Yeah, I looked at LinksysSPATFTPProv.pdf... It doesn't say, however,
how to get the phone's configuration out as a flat XML file.
Only how to push the file back into the phone.
wget
Hello all,
I'd like to thank everyone's input which I'll sumarize and comment on
bellow.
As in all complex solutions, there are no quick answers and no 100%
correct solutions. There are trade-offs to be made among
very different possiblities... Of course, the purpose of my original
On 10/9/07, Senad Jordanovic [EMAIL PROTECTED] wrote:
zoachien wrote:
Google for mexuar.
Zoa
Or look at one that works with MS Windows, Linux or Apple
http://www.bicomsystems.com/products/C/P/319/382/
FYI, Mexuar's solution -- Corraleta SDK -- *works* with
win, linux and mac, from
Hi list,
I'm evaluating a private telephony scenario of about 20
locations - 300 phones, 50 FAX machines.
Initial overview points to the installation of asterisk at three
locations connected to the PSTN via ISDN PRI.
All other locations, small by themselves, would get SIP
phones
On 9/24/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Sep 24, 2007 at 02:04:05AM +0100, Ex Vito wrote:
Is there any way to setup the asterisk cli to use such keybindings ?
...
Set in your environment:
AST_EDITOR=vi
before starting Asterisk.
(See main/asterisk.c)
Great
This might sound lika a small issu, but here it goes: I'm a long time
unix user and my shell history usage and editing is configured to use
vi keybindings; it's something that's already built into my fingers
and using different bindings, like the arrow keys to fetch previous
lines,
On 7/31/07, Jerry Geis [EMAIL PROTECTED] wrote:
I am trying to re-create calling sendDTMF in an agi and not hearing the
digit either. The above seems to re-create that without the AGI.
...you will have to configure your polycom / sip peer for inband
DTMF if you want to hear the tones.
--
David L. West wrote:
I want callers to go into the queue(s) and just hear ringing instead
of MOH. Is this possible?
...use option 'r' for the Queue application. For more options,
use 'show application queue' at the CLI.
Cheers,
--
exvito
Asterisk is loading the chan_misdn and lists mISDN when issueing show
channeltypes - however it indicates Devicestate - No. when I look for
misdn show stacks, it lists the single port of the ISDN-card, however
indicates L2Link DOWN, L1LinkDOWN. so I guess theres something wrong,
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