. Thanks!
--
*Chris Ramirez*
TELE-ONE COMMUNICATIONS, INC.
crami...@tele-onecom.com
903-531-0777
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helps here.
Chris
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Hubris Communications Inc www.hubris.net
sending all those packets in the first
place.
In other words, the qualify traffic is actually causing the problem, not
revealing it.
Chris
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New
in a
non-NAT situation and can one safely set the qualification as something higher.
I'd think something like 15 seconds would be more than enough for BLFs and
the like.
Chris
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Faisal Hanif
On 7/26/2010 9:39 PM, Chris Ramirez wrote:
The problem we are having with Asterisk is when we initiate a call
via a Zap line and it goes out on a Sip line. When it goes out via
Sip we hear no sound until the party we are calling answers the line.
If the call were to go out Sip
with the Zap-Sip calls. If anyone knows anything that could possibly
help it would be greatly appreciated. I have checked many different
things already and tried comparing Zap-Zap and Zap-Sip call logs. Thanks!
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TELE-ONE COMMUNICATIONS, INC.
crami...@tele-onecom.com
903-531-0777
about the above scenario whilst preserving the
'accountcode' field?
Thanks in advance!
Regards,
Chris
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). Is there a list of required kernel options for dahdi published
anywhere I could consult, or has anyone else come across similar errors
before?
Any suggestions gratefully appreciated.
Thanks in advance.
Regards,
Chris
/view.php?id=16705
Thoughts?
Chris
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fellows are kicking a ball around in South Africa for three
weeks, which is having an understandable effect on bandwidth usage globally.
Same happened a couple of years ago during the Olympics.
Regards,
Chris
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1.4.26.2.
Much thanks.
- Chris
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asterisk
SIP this morning they successfully
re-registered.
Is there some sort of TTL, cache, saved salt value, or other time/session
related tidbit saved that is expiring here?
- Chris
On Jun 17, 2010, at 10:21 AM, Chris Brentano wrote:
Hi all,
I tried searching, so if this has already been
I have a suspicion that it's the saved/cached SIP/IAX2 useragent info:
-- Registered SIP 'paloalto' at 10.XX.X.25 port 5060
Saved useragent Asterisk PBX 1.6.1.6 for peer paloalto
Is there a way to clear this saved info manually?
- Chris
On Jun 17, 2010, at 10:29 AM, Chris Brentano
canreinvite=no
You might also want to increase the feature code timeout (both activation
and interdigit) - I think the default is something like 500ms, which most
users find far too short to use reliably.
Regards,
Chris
it! Thanks.
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TELE-ONE COMMUNICATIONS, INC.
crami...@tele-onecom.com
903-531-0777
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,Noop( - - - - - Incoming call - - - - -)
Zeeshan A Zakaria
--
www.ilovetovoip.com http://www.ilovetovoip.com
On 2010-06-15 9:27 AM, Chris Ramirez crami...@tele-onecom.com
mailto:crami...@tele-onecom.com wrote:
We are having an issue with the RDNIS coming through with a leading 1
on some calls
are setting the RDNIS as the
CDR(userfield) to pass it through. Is that what you were wanting?
On 6/15/2010 3:23 PM, Zeeshan Zakaria wrote:
Can you give an example of how it looks like?
Zeeshan A Zakaria
--
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On 2010-06-15 4:18 PM, Chris Ramirez
.).
The eeeBox also has the advantage of being cheap (quite probably cheaper
than smaller/lower power units), which means keeping a spare around in case
of hardware failure isn't an unrealistic option.
Regards,
Chris
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really tested up to ~30 channels with
G.711 to GSM, not any of the heavier CPU workload translations (e.g. iLBC
or G.729).
For a small to medium office (e.g. 30 extensions, 10 concurrent calls) it
works fine, even with a little conferencing and transcoding.
Regards,
Chris
are within our own local calling
area. Could this be a callprogress issue and why would it only cause
problems for non-local calls?
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New to Asterisk
'
Restarting DAHDI does not help, for somehow after several hours the problem
fixes itself and I can make calls again. Any ideas on what is wrong? FWIW,
the phone line is not in use and I'm the only user.
Thanks,
Chris
On Sun, Apr 25, 2010 at 8:43 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Sun, Apr 25, 2010 at 08:37:22PM +0300, Chris Datfung wrote:
Hi,
I'm running Asterisk SVN-trunk-r226890 and whenever I restart asterisk
and
try to make a call I get the following error message
On Thu, Apr 15, 2010 at 4:06 PM, Baji Panchumarti
baji.panchuma...@gmail.com wrote:
Steve, Chris :
I too had this problem and the solution was not tweaking
the AMD parameters, but playing a short audio file (even
a really really short one) before executing the AMD function.
The key
there was nothing unique about this attack that made it require that
it come from EC2. However, that isn't true. Had this attack come from
anywhere else it would have been shut down _days_ before it was on EC2.
Chris
-
Chris Owen
On 13/04/10 00:27, Tom Stordy-Allison wrote:
Yep - this is the same codebase - the attack that I had from an EC2 yesterday
and the day before, all had the User-Agent: friendly-scanner too.
Looks like they are branching out
Go with Joshua Steins blog post - it worked perfect for me
the idea of a RBL... count me in for contributing.
I would contribute to this as well.
Chris
-
Chris Owen - Garden City (620) 275-1900 - Lottery (noun):
President - Wichita (316) 858-3000 -A stupidity tax
as
they happen. Right now everything is happening silently in the background.
Can a sound card also be used for monitoring calls even if they do not
originate from a console channel? If so, I'd appreciate it if someone could
point me in the right direction.
--
Chris
to Asterisk, but
what is a reasonable rate for each host to be allowed? This is a small
SOHO installation.
Thanks
Chris
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pointed out but it didn't make any difference.
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between GPL and Digium?
Chris
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asterisk
On 4/1/2010 1:52 PM, Kevin P. Fleming wrote:
Chris Miller wrote:
A comment in the spec file would have been nice... Does anyone know
if this a real technical issue, or simply a licensing conflict
between GPL and Digium?
It is not a technical issue; it is an issue because some of the modules
in their
network but it just seems to happen too easily and then once it stops it won't
stop.Even if this is caused by network issues is there anything I can do to
mitigate the problem. Just seems wrong that the phones would continue to ring
forever.
Chris
On Mar 10, 2010, at 9:05 PM, Warren Selby wrote:
On Wed, Mar 10, 2010 at 8:27 PM, Chris Owen ow...@hubris.net wrote:
This normally works fine but occasionally when someone picks up the call
other phones don't seem to realize the call has been answered and will
continue to ring
strangely
fascinated by this core piece of Asterisk functionality.
-Chris
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major problems with them.
Regards,
Chris
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to make sure I'm not missing something.
Thanks,
Chris
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I’ve been googling “asterisk redundancy” but all I’ve found is questions,
and no real answers.
Is this any help Dan?
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions
Chris
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Another vote for the Voiceblues. Rock solid equipment.
Peter den Hartog wrote:
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html
We use this one, and it works great.. easy to setup and it works with
a normal network connection :)
On Mon, Feb 8,
Hi
Is anyone aware of a fixed cellular terminal that supports 3G video calls?
Regards,
Chris
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Calls fail silently. There is no indication to the caller that the call is
not going through.
Has anyone else seen this?
I'd like to help debug this if I can, although I'm not sure where to start.
--
Chris
Does anyone have any suggestions as to how to make just *one* of the
DECT handsets only use the POTS but others default to their Asterisk SIP
subscriptions?
Hi Al,
I've played with the Siemens Gigaset in the past and I don't recall being
able to do this.
Chris
a print driver
on your client machines
Chris Hillman
Systems Administrator
Clearwater Research, Inc.
chill...@clearwater-research.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
. I've modified my asterisk init script to modify that
value with ulimit. My system has 192 DAHDI channels and 227 SIP peers.
When a lot of channels are in use, the number of open files climbs.
-Chris
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users
Language MOH Interpret
BlockedState
pseudodefaultdefault
In Service
What am I missing?
Thanks
- Chris
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FYI, in case anyone else encouters this issue. The card that I had
which I could reproduce this with was hardware revision B4. I RMAed
the card with Digium support and got a newer, revision C card, and the
issue is no more.
On 20 Oct, 2009, at 3:25 PM, Chris Brentano wrote:
I've seen
so. Would like to figure out what's happening here if anyone can help
shed any light as this is completely holding up migration to Asterisk
1.6 and DAHDI. Thanks.
- Chris
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://svnview.digium.com/svn/asterisk-addons/tags/1.6.1.1/
.
So just wondering if addons will be around for the forseeable future?
- Chris
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
installed a Digium TE122P in the ML350 and haven't had
any issues. I also haven't seen this in a pre-Nehalem Xeon server.
I'm using Asterisk 1.6.1.6, Dahdi 2.2.0 and LibPRI 1.4.10.1, running
on CentOS 5.3 (2.6.18-164.el5).
Has anyone seen anything similar?
- Chris
Correction, I did notice it for download at
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-addons-1.6.1.1.tar.gz
- Chris
On 16 Oct, 2009, at 4:06 PM, Chris Brentano wrote:
I noticed that asterisk.org got a redesign, quite recently it seems,
which is very nice, but the addons
Use ExtenSpy for spying on a specific extension.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExtenSpy
On 9 Oct, 2009, at 10:44 AM, Torintino T wrote:
How can i activate ChanSpy to spy on a dedicated extension?
I see the following in /etc/asterisk/extensions_additional.conf
is not conservative enough.
Just want to know the right way to handle this.
Chris
On Wed, Jul 8, 2009 at 10:35 AM, Aman Dhallyaman.dha...@live.com wrote:
Hi All,
Hope you all are fine and good, Today i have found that Mine all PRI
Channels are restating after every interval of one hour, and i have search
AIPHONE makes all that stuff, I would not try to reinvent that.
Vincent wrote:
Hello
I assume I'm not the first one to think about this: Is it possible to
connect an intercom and/or door bell to Asterisk, so that I can get an
e-mail that someone rang my place while I was out?
Even better:
= CANCEL,1,Hangup
exten = NOANSWER,1,Voicemail(${ar...@default,u)
exten = BUSY,1,Voicemail(${ar...@default,b)
exten = CONGESTION,1,Voicemail(${ar...@default,b)
exten = CHANUNAVAIL,1,Voicemail(${ar...@default,u)
exten = a,1,VoicemailMain(${ar...@default)
Chris
a call file and do this:
- exten = 5551212,1,System(/bin/cp newcall.call
/var/spool/asterisk/outgoing)
- exten = 5551212,2,hangup
Just my .02
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Mason
I have need of a very simple callback function - when any call is made
to a special SIP DID, the call is not answered but Asterisk then calls a
pre-determined number - no need for CallerID to capture the calling
number. Does anyone have a simple script to do this?
Chris
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This message has
times out on the new Asterisk server.
Is this BYE a bug in the way Asterisk handles SIP, or is this normal
SIP behavior and I should find another way to gracefully deny the call
back to Kamailio?
Thanks list,
Chris
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, then it should work as you describe.
I must confess I've never actually tried failover directly from the Snom phone
(instead using Heartbeat/LVS as you suggest), but I see no reason why it
shouldn't just work.
Regards,
Chris
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interoperability.
Regards,
Chris
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might want to specify registrar via IP rather than by name.
Regards,
Chris
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back to the previous behaviour?
Thanks in advance!
Regards,
Chris
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.
http://pastebin.com/m45e0adbd
Thanks,
Chris
On Sun, Jul 26, 2009 at 1:19 PM, Chris
Douglaschris.douglas at pioneerballoon.com wrote:
We have an inbound PRI connected to our Cisco 3825 router which is then
passing the calls to Asterisk as SIP calls. We're getting the CallerID
number
with our ADSL wholesalers, as well as with our
upstream providers (both for incoming and outbound calls). This prevents most
calls from going over an IP link where we (by which I mean us and our
suppliers, collectively) do not have end-to-end control.
Regards,
Chris
Not when you consider that there are plenty of spaces for corporate projects
as well.
On Wed, Jul 29, 2009 at 9:21 PM, Anthony antho...@rockynet.com wrote:
John Todd wrote:
What your project should have:
- No significant corporate sponsorship
JT
---
John Todd
context=ict_sip
type=friend
host=dynamic
call-limit=5
agentlogin=yes
mailbox=8...@ictvm
progressinband=no
sendrpid=yes
Any help is greatly appreciated!
Thanks,
Chris Douglas
Technical Services Manager
Pioneer Balloon Company
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for the two sides. please explan the settings and cabliing.
thanks!
Chris
On Wed, Jul 22, 2009 at 1:10 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Jul 21, 2009 at 12:01:18PM +0800, Chris YM wrote:
hello:
I wan to use the test tools-patgen and pattest for pri cards. according
?
please give me a more details in term of cables and configurations.
thanks!
Chris
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,r(myring))
Thanks
Chris
2009/6/13 David Backeberg dbackeb...@gmail.com:
On Sat, Jun 13, 2009 at 11:27 AM, Chris Maciejewskich...@wima.co.uk wrote:
Hi,
Just noticed Asterisk is not playing 'ring' tone as defined in
indications.conf when Dial command is used with 'r' option.
When I now dial
/n,60,r)
When I now dial with a SIP phone - 123 I can hear nice UK ring tone as
per [uk] definition, however when I dial 321 ring tone is different.
Is there any way to fix this?
Thanks
Chris
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in scenario B
(without using LOCAL channel)?
Regards,
Chris
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.
Regards,
Chris
2009/5/22 Martin asteriskl...@callthem.info:
it should work just fine; do you have the GSM codec compiled/loaded
core show modules like codec_gsm ... ?
OR that particular version has a BUG...
Martin
On Thu, May 21, 2009 at 3:56 AM, Chris Maciejewski ch...@wima.co.uk wrote
318
it seems that you doesent specify valid conference number
can you post meetme.conf
regards
Dhaval
On Thu, May 21, 2009 at 2:26 PM, Chris Maciejewski ch...@wima.co.uk wrote:
Hi,
I am not sure if I am doing something wrong, but I can't get MeetMe to
work with GSM codec (Asterisk
Thanks Kinjal!
Missing sound files was the problem. There were no .gsm files in my
sounds directory. Despite console shows .slin, the actual files
required are .gsm.
Once I copied .gsm into /var/lib/asterisk/sounds everything works OK.
Regards,
Chris
2009/5/22 Kinjal Dixit kinjal.di
22 10:29:34] NOTICE[6071]: chan_sip.c:7495 process_sdp: No
compatible codecs, not accepting this offer!
And asterisk is replying with 488 Not acceptable here
Any help and suggestions very much appreciated.
Regards,
Chris
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unchanged to the media tool that will use this format.
It is a media attribute, and is not dependent on charset.
Is Twinkle sending this SDP incorrectly? Or some other issue?
Thanks
Chris
2009/5/22 Kevin P. Fleming kpflem...@digium.com:
Chris Maciejewski wrote:
Found unknown media
problem, as I don't have audio files for G726?
Will try converting .pcm to .g726 and see if that will fix MeetMe issue.
Regards,
Chris
2009/5/22 Steve Howes st...@geekinter.net:
On 22 May 2009, at 16:55, Chris Maciejewski wrote:
Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer -
audio
' (language 'en')
-- Hungup 'DAHDI/pseudo-1131226973'
2009/5/22 Kevin P. Fleming kpflem...@digium.com:
Chris Maciejewski wrote:
Yes, I was missing allow=g726 for this peer :-(
Playback(/var/lib/asterisk/moh/fpm-sunshine)
works OK now, however I still can't get MeetMe to work.
Before I had
to adjust my configuration?
Regards,
Chris
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reply messages generated
by a called party?
Regards,
Chris
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by Asterisk).
2009/5/17 David Backeberg dbackeb...@gmail.com:
On Sun, May 17, 2009 at 6:43 AM, Chris Maciejewski ch...@wima.co.uk wrote:
I am trying to capture Server header in a 200 OK reply message.
My idea was to use Dail(SIP/u...@domain,30,M(GetOtherPartyInfo)),
and inside
-- INVITE [2] -- Phone 2
--- 200
OK [3] ---
What I want to do is capture Server header in 200 OK reply
generated by Phone 2.
2009/5/17 David Backeberg dbackeb...@gmail.com:
On Sun, May 17, 2009 at 9:04 AM, Chris Maciejewski ch
to
ChannelZOMBIE.
-- Executing [...@default:1] Set(SIP/somechannelZOMBIE,
CDR(DST_CODEC)=) in new stack
Is there any workaround for the above issue?
Regards,
Chris
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2009/5/17 Tilghman Lesher tilgh...@mail.jeffandtilghman.com:
On Sunday 17 May 2009 09:10:47 Chris Maciejewski wrote:
Hi,
I am using SHARED() function to push destination channel info (i.e.
audio codec) into source channel, in order to record into a customer
CDR field.
My dialplan looks like
into an Asterisk
spool directory and a call will be placed
* Asterisk Manager API Use the Originate command
Regards,
Chris
2009/5/13 Dan Caescu dcae...@eqnet.us:
Forget the typo (s/ANSWERED/ANSWER/g)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk
of the phone, without the aid of a 3rd
party VOIP app.
Any thoughts on the 4G spec and its influence on the future of
Asterisk dev?
Thanks,
Chris
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,
Chris
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What would be the correct application/function to generate 404 Not found?
Thanks for help,
Chris
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is needed to do this, is to use correct 'causecode' as Hangup
parameter :-)
exten = i,1,Playback(you-dialed-wrong-number,noanswer)
exten = i,n,Playback(check-number-dial-again,noanswer)
exten = i,n,Hangup(1) ; - NOTE: causecode 1 for 404 Not found
Best regards,
Chris
Yes, 'causecode' parameter of Hangup application was missing at:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Hangup
I have added 'causecode' to the above wiki page now.
Thanks for your help,
Chris
2009/4/16 Tilghman Lesher tilgh...@mail.jeffandtilghman.com:
On Thursday 16 April 2009 10
to have a habit of introducing hum into otherwise perfect calls.
As an aside, one does wonder why there's such inadequate shielding against
interference on SIP phones, given that most environments in which they'll be
used will also have a cellphone on the desk.
Regards,
Chris
, it might be worth considering that instead if you don't need ISDN or
POTS connectivity.
I've done a few Asterisk-based eeeBoxes over the last few weeks and been very
impressed with them.
Regards,
Chris
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wallet in a big way when your phone bill comes round.
Thanks to everyone in advance.
Regards,
Chris
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]
Sent: 27 March 2009 4:36 pm
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] London DDI test request
Quoting Chris Bagnall li...@minotaur.cc:
Greetings list,
I'm trying to establish if there's an issue whereby certain telcos
in certain countries have not updated
tried on of
these - but they look good (and can contain a camera as well if needed).
HTH
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris
Mason (Lists)
Sent: 17 March 2009 13:23
To: Asterisk Users
Hi,
We have a PRI Trunk (physical E1) and we are getting
some rather weird and very isolocated problems. On outbound calls to
specific numbers, it would seem to me that DTMF from the remote side is
affecting the local asterisk system. Basically what happens:
- We make a OUTBOUND call via the
Has anyone found a good wayt o do a gate intercom using Asterisk? I am
looking at a Xorcom PBX with programmable contact, so I have no issue
with opening the gate, but the interface at the gate is a bit tricky. I
thought about a weather proof housing containing a phone but it seems a
bit
I'm not getting inbound audio from bandwidth.com. Their engineer said the
invite that they're sending me looks like this:
INVITE sip:+15129616...@67.198.16.18:5060;transport=udp SIP/2.0.
Record-Route: sip:216.82.224.202;lr;ftag=VPSF506071629460.
Record-Route:
4.68.250.148:5060;branch=z9hG4bK506071629460-1207517079314
From: GARRIGUES,CHRIS
sip:+15124990...@4.68.250.148sip%3a%2b15124990...@4.68.250.148
;isup-oli=0;tag=VPSF506071629460
To: sip:+15129616...@4.79.212.229:5060
Call-ID: houmgc0520090316174121064...@209.244.63.35
CSeq: 1 INVITE
Contact: sip
Hi,
Is there any way to tell Asterisk not to generate additional headers like:
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
I can't find any relevant option in sip.conf file :-(
Thanks for help.
Chris
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something else.
2009/3/15 Olivier oza-4...@myamail.com:
2009/3/15 Chris Maciejewski ch...@wima.co.uk
Hi,
Is there any way to tell Asterisk not to generate additional headers like:
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
I can't find any relevant option
to
blame but ourselves.
As others have said, there are opportunities in the the current financial
climate that wouldn't otherwise be available.
Regards,
Chris
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