Hi all
I have posted a question on the asterisk dev board about this issue but I
want to see if any users have run up against this.
This issue is that when calls are run through Broadvox and Level 3 the
in-call rfc2833 dtmf is not reliable. This occured for me on asterisk
version 1.6.1.18,
Hey all
We are seeing intrusion attempts coming from address 201.47.236.122 today
They were hitting our switches trying to get in. So we blocked them at our
firewall.
Just wanted to put the word out so you all can protect your self.
Bryant
--
From: Bruce Ferrell bferr...@baywinds.org much static testing of my
realtime configuration and applications I'm
almost ready to pull the trigger.
The one thing I've been able to determine is what I need to do to
migrate my g729 licenses.
Has anyone got any advice for me on this? The Digium
I have found it best when doing remarks to not use the ;- combination as I
have seen it cause failuers on dialplan reload.
Bryant
What I saw was that Asterisk stumbles when putting a comment like this :
;-- bla bla !!!
It should be :
; -- bla bla !!!
So with a space between ; and --
The
On Sat, Aug 28, 2010 at 3:22 AM, Jonas Kellens jonas.kell...@telenet.be
wrote:
Hello list,
I have a file to be played in wav-format.
I thought Asterisk would automatically take the wav-file and translate it
to the codec used, but I see this :
[Aug 28 11:16:29] WARNING[2705]: file.c:664
Todd
How do you have the context in the phones sip configs set?
Bryant
From: Todd Reese trees...@gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on
=0.0.0.0/0.0.0.0
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/150
context=from-trunk
canreinvite=no
callgroup=
callerid=device 150
accountcode=
call-limit=50
On 8/30/2010 10:37 AM, Bryant Zimmerman wrote: Todd
How do you have the context in the phones sip configs set?
Bryant
From: Todd Reese
On blind transfers I believe the two cdr's have the same unique id . On
attended transfers there is no real way I have found to address this issue.
CDR's with transfers really suck the way they are right now. On blind transfers
you can do some flagging of the second CDR by checking in your
Nic
How stable is 1.8 really? It sounds like you are running it in production is
this the case? CDR Transfer issues and rfc2833 DTMF issues are hitting us hard
with 1.6.2.x. We want to move as soon as 1.8 is stable enough.
Thanks
Bryant
From: Nic
Richard
Who is the carrier that the calls are flowing in from?
Bruamt
From: ken...@gnat.com (Richard Kenner)
I'm having a wierd problem. Somewhere around 1-2% of the time, the
first DTMF digit dialed gets dropped. This is occurring during a
I have seen simalar issues with some audiocodes gear. I adjusted the early
media options on the pri in the audiocodes gateway and that fixed my
issues. We have also seen this when calls come in from Level 3 toll free
some times their gatways screw with things. We found adding an manual
answer
The issue we are having is that in-call RFC2833 DTMF digits are being
dropped with Broadvox and Level 3. This is happening with Grandstream GXP
and Snom phones. We did some testing with the vendors and here is one of
the responses we got back. Is there any way to force asterisk to modify the
Is there a way to drop a ip connection to asterisk after a number of
register attempts.
I have been having issues with hackers doing registration scanning against
our server. We block their address at the fire wall but since asterisk does
not force a drop of the connect after so many bad reg
Steve
Grandstream has a new services GXP-21XX coming out they may work for your.
We have been a beta tester and the BLF on these seem to work much better
then the GXP-20XX units. I do not have the side cars in stock right now so
I don't know how they work with it but you can put at least two
As I look to move our systems to version 1.8 I am looking at making a
change from mySQL to PostgreSQL.
I love mySQL but am getting very concerned about i'ts new owners.
Should I be able to move all my realtime stuff to PostgreSQL is it fully
supported with asterisk?
Is there any down side to
Is there the ability in the Asterisk 1.8 CEL logging to log the SIP
endpoint IP as weell as the medie enpoint's ID's?
Thanks
Bryant
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New to Asterisk?
We can resell the Sangoma card. They have some higher license counts as
well.
They are also offering a step up offering. If you buy at one level and need
to move to the next.
They will offer you a trade back on the old card.
Bryant
From: Tim Nelson
On 09/23/2010 06:48 AM, Tarek Sawah wrote:
Greetings,
Because of the heavy load and the high expectations of an asterisk
server
offered as a solution integrated with our CRM software.. we were looking
into other possibilities than software Licenses for G729 and G723
codecs..
to lower the
Version 1.6.2.13 is having issues with audio prompts dieing. When users
call in to get voicemail the prompts start and then stop about 6 to 10
seconds in. On hold music plays for 6 to 10 seconds and then stops. In meet
me conference rooms hold music will stop about 6 to 10 seconds in. Audio
Hi All
In the console I am seeing warring rtp.c:1635 ast_rtp_read: RTP Read too
short
I get these all of the time things seem to be working fine but I am trying
to figure out if there is a way to resolve these Warnings.
I am running asterisk 1.6.2.13
Any direction is appreciated.
Thanks
Tim
I am actually seeing this on a 1.6.2.13 box as well. For some reason
durring prompt playbacks they some times stall mid file. The call stays up
but no audio comes in.
Bryant
From: Tim Panton t...@westhawk.co.uk
Sent: Friday, October 08, 2010 10:38
On 8 Oct 2010, at 15:37, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton
Sent: Friday, October 08, 2010 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of
the three perform well in all enviroments. Between stablity issues, T38 and
DTMF talkoff all three suffer some combination of issues.
I am looking at Patton and Innomedia. Has any one tried either brand and
what is your
Us too. Tons of SPA2102's out there working fine!
On Fri, Oct 8, 2010 at 4:36 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Fri, 8 Oct 2010, Bryant Zimmerman wrote:
I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of
the three perform well in all
enviroments. Between
Anyone used the new Grandstream GXP-21XX series phones. We have been
testing these phones and like what we see. We are looking for a greater
cross section of testing before we roll them to production. Any feed back
would be appreciated. We are talking with Grandstream engineering and they
are
We are testing the innomedia ATA's to possibly replace our current line up
of ATA's that we are using. Has anyone used their product? What is their
track record on stability, voice quality, DTMF talkoff, T.38
Thanks
Bryant
From: Zeeshan Zakaria
@lists.digium.com
Subject: Re: [asterisk-users] GXP-21XX
On Wed, 13 Oct 2010, Bryant Zimmerman wrote:
Anyone used the new Grandstream GXP-21XX series phones. We have been
testing these phones and like what we see. We are looking for a greater
cross section of testing before we roll them
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias
Sent: Wednesday, October 13, 2010 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] checking CDR
Hello
The real question is are you having the phone forward the calls or is your
dial plan redirecting to outbound calling?
Bryant
From: Zeeshan Zakaria zisha...@gmail.com
Sent: Wednesday, October 13, 2010 2:16 PM
To: Asterisk Users Mailing List -
From: Paul Belanger paul.belan...@polybeacon.com
Sent: Thursday, October 14, 2010 5:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Audiocodes firmware
On Thu, Oct 14, 2010
We are being forced to move away from audiocodes ATA's because they refuse
to fix a few minor bugs unless we commit to a 1000 piece order. This is on
their 2 port ATA's. Their response to us is that ATA's are intended for
serious carriers that are using them in conjunction with their higher end
When we designed our systems on asterisk we designed it to me multi-tenant.
Se we use customer prefixes on all extensions. This allows us to have
multiple customers using the same extension pools. It also reduces the hack
foot print as hackers must know the prefix for a customer to try and
I would look at x10 triggered switches. There are some command line tools
you could call from an IVR.
I did a lot of x10 development on windows back in the day. I have seen some
things for linux as well.
http://www.heyu.org/
Bryant
From: C F
We are having issues with asterisk 1.6.2.12-rc1 and 1.6.2.13 with audio
playback randomly stopping during calls.
A caller goes to voice mail and the prompts stop playing back. IVR prompts
stop playing in mid stream. This occurs randomly and is causing quite a
problem. I do not see any errors or
Bria is a full SIP soft client. It works ok if you have a very good sound
card and good wired headset.
It is not a dialer application in the sense that you would dial your desk
phone using it.
Some of my clients love the Bria and some say the quality is poor. You must
have a computer that can
I have used 1.4 1.6. I am testing 1.8 for production and it is looking
very good. I am making some changes to accommodate some minor dialplan
changes from 1.6. Our 1.4 is very solid 1.6 has some issues with DTMF
issues when used with Sonus on the back end. 1.8 is looking very good and
we hope
From: Mark Scholten m...@streamservice.nl
Hello,
I have a few questions regarding Asterisk 1.8.0. If you can answer a
question, please do so.
Is Asterisk 1.8.0 stable enough for production environments?
It appars to be so far we are testing and
I am having issues with Blind Transfer on asterisk 1.8
If I call from one Grandstream phone to another and us the transfer key
to do a blind transfer everything works fine.
When calling in on a sip trunk and then trying to use the transfer key
to transfer from Grandstream phone to
Replys from Bryant
On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com
wrote:
I am having issues with Blind Transfer on asterisk 1.8
What specific version: 1.8.0, 1.8.1-rc1, SVN branch? What OS?
Verison 1.8.0, Suse 11.1
If I call from one Grandstream phone to another
Subject: Re: [asterisk-users] Issues with 1.8 and BlindTransfer
Hi,
Am Donnerstag, den 02.12.2010, 11:02 -0500 schrieb Bryant Zimmerman:
Replys from Bryant
On Wed, Dec 1, 2010 at 9:44 AM, Bryant Zimmerman brya...@zktech.com
wrote:
I am having issues with Blind Transfer on asterisk 1.8
What
Nikhil
Known bug. there is a patch that is in the SVN trunk. I just downloaded the
trunk version last night and will be testing in a bit.
I will keep you posted.
Bryant
From: Nikhil d.nik...@cem-solutions.net
Sent: Monday, December 06, 2010 6:41 AM
To:
I belive the WBP54g cisco/LINKSYS adapter is what we are using with the
Grandstream phones. You have to buy a Cisco/Linksys power supply but it
works great. I have over 200 of them out there.
Bryant
From: Jeremy Betts jer...@freevoicepbx.com
Sent:
I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql
table for CDR's today there are no entries since the update.
I have rebuilt and re-installed and re-started asterisk still no CDR's
flowing to mysql. I did not change any configs. I checked to make sure that
the
I am trying to include the ${HANGUPCAUSE} in my mySQL cdr tables. I have a
field called cause_code but it won't write. I belive it is because the
record has already been written by the time I hit the h section of the
code. How might I get this info into the CDR. I need this info for Quality
I see the same thing. Why is there an CANCEL status if it is never set. The
only way I have been able to capture a Cancel status is with the
h extensions using the 'e' option under dial. But this leaves no way to
tell what the DIALSTATUS state was as it is blank. I belive it is a bug as
well.
To my knowledge there is currently no free version of the g729 codec. There
were some spec builds but those were just for testing if I recall
correctly. Each version of the codec that we have always gotten has been
compiled for each version of asterisk. I would just buy the Digium licenses
The Dial Status is not set when accessing it from the h extension.
Bryant
From: Vardan Harutyunyan hvarda...@gmail.com
Sent: Wednesday, December 22, 2010 10:39 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DIALSTATUS on CANCEL
Giorgio
You could buy just a couple of licenses 3 to 5. It would get rid of the
messages for the most part and it would give you the ability to transcode
for voicemails and other items requiring transcode. The reason you are
likely getting the messages is there is some kind of transcode
Ok I can't get my CDR values to set from the h extension in either 1.6.2 or
1.8 What is wrong? Here is what I found in the cdr.conf
; Normally, CDR's are not closed out until after all extensions are
finished
; executing. By enabling this option, the CDR will be ended before
executing
; the
What would it do if you
exten = h,1,ResetCDR(w)
exten = h,2,NoCDR()
exten = h,3,DEADAGI(get-unqiueid.php)
I have not tried it but in theory it should write the first CDR and then
kill the write of the second NO ANSWER CDR.
Let me know if it works for you as I may need to do it on some of my h
-users@lists.digium.com
Subject: Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wednesday 22 December 2010 11:42:33 Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from the h extension in either 1.6.2
or 1.8 What is wrong? Here is what I found in the cdr.conf
From: Carlos Chavez cur...@telecomabmex.com
Sent: Wednesday, December 22, 2010 2:46 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wed, 2010-12-22 at 12:42 -0500, Bryant Zimmerman wrote:
Ok I can't get my CDR values to set from
: Thursday, December 23, 2010 12:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Possible Bug (Include ${HANGUPCAUSE} in CDR)
On Wednesday 22 December 2010 21:08:56 Bryant Zimmerman wrote:
My h extension is in the same
Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Bryant Zimmerman wrote:
The Dial Status is not set when accessing it from the h extension.
Bryant
dbackeb...@gmail.com
Sent: Thursday, December 23, 2010 10:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] cdr_mysql stopped working
On Mon, Dec 20, 2010 at 5:02 PM, Bryant Zimmerman brya...@zktech.com
wrote:
I did
records.
David Backeberg wrote:
On Mon, Dec 20, 2010 at 5:02 PM, Bryant Zimmerman brya...@zktech.com
wrote:
I did an upgrade to the SVN trunk on the 12/9 and when I looked in my
mysql
table for CDR's today there are no entries since the update.
I have rebuilt and re-installed and re-started
I am setting up CEL with asterisk 1.8 and so far so good. The issue I was
hoping to address here was also being able to get storage of other values
such as HANGUPCAUSE and other variables that are used for billing and
quality of service. The CEL documentation starts out by saying that we can
How can I get the media and sip endpoints IP address durring h
extension?
I need to write these to my CEL logs.
Any ideas?
Thanks
Bryant
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
If you are using CEL in asterisk 1.8 can you please look at the issue
tracker and comment.
On how this might effect you.
https://issues.asterisk.org/view.php?id=18559
Thanks
Bryant
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-- Bandwidth and Colocation Provided by
Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Bryant Zimmerman wrote:
Vardan
I have not use AEL so it is a bit hard to follow with the formatting
the
way
Use a combination of ${EPOCH} with a format string and the unique call /
channel id.
Example:
exten = s,1,Set(MY_TIMEVAR=:${STRFTIME(${EPOCH},,%d%mNaVH:NaVS)})
exten = s,n,Monitor(wav,${MY_TIMEVAR}~${CHANNEL},m)
From: bilal ghayyad
)
exten = rfc-3325-CPN,n,Return()
Good Luck
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003 Good LuckBryant Zimmerman (ZK Tech Inc.)616-855-1030
Ext. 2003
From: Jonas Kellens jonas.kell...@telenet.be
Sent: Wednesday, January 05, 2011 9:44 AM
Jonas
What is the dtmf setting on all peers involved in the call?
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: Jonas Kellens jonas.kell...@telenet.be
Sent: Wednesday, January 05, 2011 4:55 PM
To: Asterisk Users Mailing List
From: William Stillwell will...@stillwellsoft.com
Sent: Thursday, January 20, 2011 11:26 AM
This is new to me, I have a fax server using Receive Fax and gets way over 5
calls at a time. [fax-in] exten = s,1,Answer() exten = s,n,Wait(1) exten
=
On 01/20/2011 11:47 AM, Steve Underwood
On 01/20/2011 11:11 PM, Kevin P. Fleming wrote:
On 01/19/2011 02:30 PM, Bryant Zimmerman wrote:
On 01/19/2011 02:05 PM, Bryant Zimmerman wrote:
I am working on some fax tools for some of my users. I am reading the
https://wiki.asterisk.org docs
faxes
calls are then sent to either 1.6.x or 1.8.x boxes and then on to the final
ata.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: Amit Nepal ami...@phoenixinternet.net
Sent: Thursday, January 20, 2011 4:27 PM
To: asterisk-users
Where can I get the latest stable version of spandsp. That will work with
res_fax_spandsp.so. The link listed on the voip-info website is dead. Any
other location for download?
http://www.soft-switch.org/
Thanks
Bryant Zimmerman
On 01/24/2011 2:54PM Bryant Zimmerman wrote
I am testing out inbound faxing using res_fax and res_fax_spandsp.so My
system answers the call but then sets there on the ReseiveFax line then
comes back with an error that it exceeded the maximum retries. How
PM, Bryant Zimmerman brya...@zktech.com
wrote:
Do you know how to force off T.38 in res_fax?
it's in sip.conf
take a look for
t38pt_udptl=yes
change it to no
reload sip
on your console
that should force it to either fail entirely or do audio passthrough
Has anyone else seen an issue with t.38 faxing on Level 3 with res_fax and
res_fax_spandsp.so
What we are seeing in the packet captuers is that the call is trying to do
t.38 but does not appear to be completing the handshaking. No data is being
transmitted. I have included a link to my pcap of
From: viswavardhanreddy karna viswavardhanre...@gmail.com
Sent: Wednesday, January 26, 2011 11:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] Regarding error in Asterisk dail
Steve
Are there any undocumented options available with ReceiveFAX and the
res_fax_spandsp module.
I am having issues with getting t.38 to negotiate with Level 3 faxes but if
I force t.30 the fax comes in. But the fax does not fall back t.30 if the
t.38 fails
Thanks
Bryant Zimmerman (ZK
From: Kevin P. Fleming kpflem...@digium.com
Sent: Wednesday, January 26, 2011 1:50 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/26/2011 12:42 PM, Bryant Zimmerman wrote:
Steve
Are there any undocumented options
From: Kevin P. Fleming kpflem...@digium.com
Sent: Wednesday, January 26, 2011 2:29 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/26/2011 01:19 PM, Bryant Zimmerman wrote
From: Kevin P. Fleming kpflem...@digium.com
Sent: Wednesday, January 26, 2011 4:52 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/26/2011 03:14 PM, Bryant Zimmerman wrote:
Is there a way for me to force t.38 off
From: Kevin P. Fleming kpflem...@digium.com
Sent: Wednesday, January 26, 2011 5:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/26/2011 04:16 PM, Bryant Zimmerman wrote
Kevin
That is grate. I dove into the code and tried to add it my self I added
a F option but I have not figured out the right spot to force the
exclusion to shut off the T38.
Where will the patch be posted?
http://svnview.digium.com/svn/asterisk?view=revrev=304342
Kevin
That is grate. I dove into the code and tried to add it my self I added
a F option but I have not figured out the right spot to force the
exclusion to shut off the T38.
Where will the patch be posted?
http://svnview.digium.com/svn/asterisk?view=revrev=304342
Kevin
I tried
From: Kevin P. Fleming kpflem...@digium.com
Sent: Thursday, January 27, 2011 10:31 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/27/2011 09:21 AM, Bryant Zimmerman wrote:
Kevin
That is grate. I dove into the code
From: Kevin P. Fleming kpflem...@digium.com
Sent: Thursday, January 27, 2011 3:08 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/27/2011 09:21 AM, Bryant Zimmerman wrote:
Kevin
That is grate. I dove into the code
From: Kevin P. Fleming kpflem...@digium.com
Sent: Thursday, January 27, 2011 3:08 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/27/2011 09:21 AM, Bryant Zimmerman wrote:
Kevin
That is grate. I dove into the code
From: Kevin P. Fleming kpflem...@digium.com
Sent: Monday, January 31, 2011 5:13 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] res_fax
On 01/31/2011 02:08 PM, Bryant Zimmerman wrote
this with the
1.6.x builds.
Is there a way to reload the hints or force a refresh without re-starting
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
--
_
-- Bandwidth and Colocation Provided by http://www.api
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Tuesday, February 15, 2011 1:16 PM
To: Asterisk Users List
Subject: [asterisk-users] Voicemail email attachment as
I had an issue today where receive_fax caused an asterisk switch to crash.
The switch is 1.8.2.3 version. The call was coming from a fax machine. The
call started receive_fax answered and then asterisk stopped responding. I
was able to log into asterisk but it would not do a core restart now nor
From: Jonas Kellens jonas.kell...@telenet.be
Sent: Wednesday, March 09, 2011 4:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] SIPAddHeader not working
Hello list,
I notice
From: --[ UxBoD ]-- ux...@splatnix.net
Sent: Wednesday, March 09, 2011 6:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] Multiple SIP endpoint registrations
Hi,
With Asterisk
From: satish patel satish...@hotmail.com
Sent: Tuesday, March 15, 2011 2:31 PM
To: asterisk-users asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call file for page auto-call
Thanks for you input but how to do SIPAddHeader(Alert-Info:
From: Bob Beers bob.be...@gmail.com
Sent: Friday, March 25, 2011 10:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk 1.8 question
On Fri, Mar 25, 2011 at 9:51 AM,
From: Jonathan Thurman jonat...@thurmantech.com
Sent: Friday, March 25, 2011 11:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Why shouldn't I use 1.8?
On Fri, Mar 25,
calls. If
you use it right it is rock solid.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: Rizwan Hisham rizwanhas...@gmail.com
Sent: Monday, April 04, 2011 12:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk
I have deployed several 1.8.3.2 systems as upgrades of customers systems
and now I am seeing random crashes. For some reason the builds lock up and
stop taking sip connections. Existing calls stay on but when the user hangs
up no new calls or reg attempts work. In most cases a core restart now
On 4/5/11 6:10 PM, Bryant Zimmerman wrote:
I have deployed several 1.8.3.2 systems as upgrades of customers systems
and now I
am seeing random crashes. For some reason the builds lock up and stop
taking sip
connections. Existing calls stay on but when the user hangs up no new
calls or reg
On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@officegeneral.com
wrote:
On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
Thanks for your response. I have added the patch for 18818 per
Michel Verbrask's
recomendation. It appers that it has made quite a difference. I
don't have an PRI
On Apr 7, 2011, at 8:51 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
On Thu, 2011-04-07 at 08:37 -0400, Bryant Zimmerman wrote:
On Apr 6, 2011, at 8:54 PM, Edwin Lam edwin@officegeneral.com
wrote:
On 4/6/11 3:02 PM, Bryant Zimmerman wrote:
Thanks for your response. I have added
From: Chris Owen ow...@hubris.net
Sent: Thursday, April 07, 2011 9:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk 1.8.3
Best I can tell, multi-tenant parking also
Is there a way to place a hangup time on a dynamic Meetme conference. I am
using Page() with a Meetme conf and I have had a few instances where
someone from a wifi voip phone looses ip while doing a page and the page
never hangs up. I have to kill it. I need to somehow limit the page so
after
I will throw in my 2 cents on this. I agree that 1.8 is not as stable as it
needs to be. From my perspective I will continue to use the 1.4.x or
1.6.2.x release that is the best fit for me and it should continue to do
what it does and it get's it's security releases.
If the primary development
I have been using sendfax and recievefax with 1.8.x.x version I have a
patch that Kevin Fleming wrote to allow the forced shutoff of T.38 F
option. This was considered a new feature so it is not in new releases of
1.8.x and I have not been able to get a patch working for the current
releases.
From: Ira i...@extrasensory.com
Sent: Thursday, May 05, 2011 12:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4
behind?
At 07:56 AM
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