Hi all,
For the most part, we are finding that Fax for Asterisk works pretty
well. However, we have seen some wierdness that we'd like to try to
fix.
Once in a while, we will get a partial result report for a given fax
but when we look at the actual .tiff image, it looks to be complete.
This is
On 03/11/2014 12:36 AM, Mike Diehl wrote:
Hi all,
For the most part, we are finding that Fax for Asterisk works pretty
well. However, we have seen some wierdness that we'd like to try to
fix.
Once in a while, we will get a partial result report for a given fax
but when we look at the actual
One of my customers had this problem. The cause turned out to be the other side (one particular
remote station).
In addition to res_fax, I also installed the Hylafax/iaxmodem combo. Hylafax has the advantage
of giving nice logs about the signaling. This way it was easy to see what was going
Hi
Le 10/03/2014 17:36, Mike Diehl a écrit :
Hi all,
For the most part, we are finding that Fax for Asterisk works pretty
well. However, we have seen some wierdness that we'd like to try to
fix.
Once in a while, we will get a partial result report for a given fax
but when we look at the
11.7 is working fine for me.
I put on 11.8 and my confbridge that should be muted
to users now sounds un-muted or like I am getting feedback.
Did something change there???
Jerry
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Steve,
I BELIEVE the fax is complete because the fax image is a form that appears
to only be a single page.
But, since FFA isn't providing acknowledgement, the sending fax machine is
resending the document multiple times!
Mike.
On Mon, Mar 10, 2014 at 12:49 PM, Steve Underwood
Guys, hi. I have not solved the problem. Outgoing calls to remote
extensions drops on 5-20 seconds. Incoming calls work perfectly.
Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq
Thanks,
On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling ewiel...@nyigc.com wrote:
See
Try ulaw instead of g729, set directmedia=no
I see you are using FreePBX. I cannot help further.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com
Sent: Monday, March 10, 2014 4:15 PM
To:
Yes, well, really is Elastix. Hmmm where I need to pt directmedia=no ?
Thanks,
On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling ewiel...@nyigc.com wrote:
Try ulaw instead of g729, set directmedia=no
I see you are using FreePBX. I cannot help further.
-Original Message-
From:
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security
releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1,
and 12.1.1.
These releases are available for immediate download at
Asterisk Project Security Advisory - AST-2014-002
ProductAsterisk
SummaryDenial of Service Through File Descriptor Exhaustion
with chan_sip Session-Timers
Asterisk Project Security Advisory - AST-2014-001
ProductAsterisk
SummaryStack Overflow in HTTP Processing of Cookie Headers.
Nature of Advisory Denial Of Service
Asterisk Project Security Advisory - AST-2014-003
ProductAsterisk
SummaryRemote Crash Vulnerability in PJSIP channel driver
Nature of Advisory Denial of Service
Asterisk Project Security Advisory - AST-2014-004
ProductAsterisk
SummaryRemote Crash Vulnerability in PJSIP Channel Driver
Subscription Handling
On Fri, Mar 7, 2014 at 12:38 AM, hkc323 hkc...@gmail.com wrote:
===
Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
Program terminated with signal 11, Segmentation fault.
For the developers who can interpret an Asterisk
Check here:
http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0
Thanks,
Steve Totaro
On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com alp...@gmail.com wrote:
Yes, well, really is Elastix. Hmmm where I need to pt directmedia=no ?
Thanks,
no trunking or bonding involved, so why just everybody calls this a trunk?
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On 03/10/2014 07:39 PM, Thomas Rechberger wrote:
no trunking or bonding involved, so why just everybody calls this a trunk?
It is just another SIP peer. You tend to route more than one extension
down/from it.
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Because sometimes marketing overcomes technical correctness.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Rechberger
Sent: Monday, March 10, 2014 7:39 PM
To: asterisk-users@lists.digium.com
Hi Mike,
If the sending machine keeps trying it might be the call has been hung
up by asterisk before its own acknowledgement message has finished being
sent. There have been bugs like this in the past, and people can be
pretty casual about making changes which hang up aggressively. A FAX
Hello Everyone,
I am using this bash script to pull resource id
http://fpaste.org/84173/51169913/
this whole macro http://fpaste.org/84174/51176013/
that what I see when dialplan ran
Executing [s@macro-missed-call-in:3] Set(SIP/babytel-0022, RES=9c32ecc4
-- ) in new stack
-- Executing
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