=dynamic
description=Polycom
context=sip
qualify=yes
rtptimeout=60
rtpholdtimeout=60
rtpkeepalive=60
callerid="Polycom "
qualify=no
canreinvite=yes
timezone=1
nat=force_rport,comedia
disallow=all
allow=ulaw
allow=alaw
allow=gsm
Thoughts on what is happening here or wha
me since
last read = 0ms), dropped 5838 packets
On 02/13/2018 01:24 PM, Andres wrote:
On 2/13/18 11:55 AM, Michael Maier wrote:
On 02/13/2018 at 08:41 AM Floimair Florian wrote:
No you're reading it wrong.
There are 188K received wi
(time since
last read = 0ms), dropped 5838 packets
On 02/13/2018 01:24 PM, Andres wrote:
On 2/13/18 11:55 AM, Michael Maier wrote:
On 02/13/2018 at 08:41 AM Floimair Florian wrote:
No you're reading it wrong.
There are 188K received with no loss, and 16441K transmitted.
This doesn't make
On 2/13/18 11:55 AM, Michael Maier wrote:
On 02/13/2018 at 08:41 AM Floimair Florian wrote:
No you're reading it wrong.
There are 188K received with no loss, and 16441K transmitted.
This doesn't make any sense to me, either. There can't be more packages
transmitted than received. It's the
ing on these other ports.
Together with the allow/deny pjsip settings, I *think* I'm reasonably safe?
What bothers me is that don't understand how/why this is happening.
And that makes me nervous!
Thanks.
--
Andres
--
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On 1/12/17 11:09 AM, Telium Technical Support wrote:
This was asked many years ago but I thought I would check to see if
things have changed. Is it possible to take over a call in progress –
using a replacement Asterisk server?
One plausible scenario I can think of is if you are running
Hi,
Consider the following scenario. A customer's incoming call enters the
system, and after some processing, the call is placed on a queue, where it
will be picked up by an agent.
Then, the agent makes an attended transfer (using asterisk internal
transfer) of this costumer to some other
On 3/31/16 7:34 AM, Roel van Meer wrote:
Hi list!
I have a problem where SIP packets sent by Asterisk do not hit the
wire, and I don't know what could cause this.
I'm running Asterisk 1.8.28_cert5 with full SIP debug. At the same
time, I'm doing a tcpdump of the traffic on the network
) Is a 'ds_select_dst()' followed by a 'forward()' the right way to
route calls in OpenSIPS? It works most of the time.
2) Can (or should) I configure Asterisk to not send the INVITE at 15
minutes?
On Sat, 21 Nov 2015, Andres wrote:
Looks like session timers are kicking in and a Re-Invite is being
sent
On 11/20/15 11:13 AM, Steve Edwards wrote:
I have a problem where SIP calls from some providers are dropping at
15 minutes.
The topology is: Client sends calls to a host running OpenSIPS,
OpenSIPS sends calls to an Asterisk server.
Below,
'Client' is the IP address of the client's host
On 9/13/15 11:16 AM, Gokan Atmaca wrote:
Hello
I'm using the Fail2ban. I configuration below. I want to try to
prevent the continuous password. Fail2ban password that does not
prevent this form. (Asterisk 1.8 / Elastix interface)
What could be the problem ?
Asterisk log;
"Registration from
On 8/21/15 6:45 PM, Technical Support wrote:
I got a new SNOM M65 which works fine for outgoing calls, but incoming
calls never ring at the handset. I captured the SIP traffic and see
that my M65 is replying with an 488 not acceptable here. From what I
read this is usually codec related but
On 7/22/15 1:38 AM, Brendan Ord wrote:
I’ve gotten to the bottom of this;
Seems that the pjsip.endpoint_custom.conf isn’t getting included
properly, or my syntax is wrong.
Last time I checked you have to put a plus sign to combine parameters
from main and custom file. Like this:
On 7/6/15 5:53 PM, Jamie Rees wrote:
Hello folks,
We have an issue with several Cisco SPA512G phones connected to an
Asterisk platform where several users hear loud, random beeps during
calls to external recipients. The noises are akin to button press
tones, are very loud and a significant
On 6/8/15 1:18 AM, Luca Bertoncello wrote:
Hi list!
Another day, another problem...
I'm checking with Nagios my Asterisk at home, and since yesterday I noticed
that, after my IP changes (Deutsche Telekom drop the DSL-line every 24 hours,
so that I have a new IP every day), the peer of an
On 4/7/15 7:48 PM, Andrew Galdes wrote:
Hi Dmitriy and others and thanks for your help so far.
The option match_auth_username=yes seems to have had no effect. From
my reading, this option will try to match the username of the incoming
SIP account to a section heading. If that is how it must
with the restriction of the firewall that should be a
secure solution.
Am 01.04.2015 um 19:23 schrieb Sebastian Kemper
sebastian...@gmx.net mailto:sebastian...@gmx.net:
On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote:
On 4/1/15 10:48 AM, Daniel Heckl wrote:
John
On 4/1/15 10:48 AM, Daniel Heckl wrote:
John,
thank you four your answer. I think you have misunderstood the
problem. It’s about a ip address change of the sip trunk, not of my
asterisk server.
You would probably benefit by enabling the DNS Manager to allow for
dynamic IP changes:
# cat
On 4/1/15 7:50 PM, Andrew Galdes wrote:
Hello all,
I have an Asterisk server (Asterisk 10.12.4) with multiple sip
accounts with the same service provides. We have 8 phone numbers in
total.
Incoming calls from the public are all correctly directed to
appropriate office handsets. However,
On 3/12/15 9:39 AM, Ron Wheeler wrote:
Your characterization may be true but Skype works much better than SIP
when it comes to sound quality.
SIP is not to blame for this. Its the audio codec being used. Skype has
spend a great deal of effort with their SILK codec by making it highly
On 2/20/15 6:15 AM, thufir wrote:
What's the difference between user 123 and devries? Based on the
output here, they seem the same..?
tleilax*CLI
tleilax*CLI sip show users
Username Secret Accountcode
Def.Context ACL Forcerport
201
On 2/20/15 3:20 PM, thufir wrote:
On Fri, 20 Feb 2015 15:07:35 -0500, Andres wrote:
This is showing nothing so I don't think your test message even made it
here. I think it looped in the 'doge' server.
I was wondering the same thing :)
in tleilax, I looked in /var/log/asterisk/messages
On 2/20/15 2:29 PM, thufir wrote:
On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote:
A sip set debug on will give you more info on why you are getting the
404. It probably has to do something with your context/dialplan.
on tleilax:
tleilax*CLI
tleilax*CLI sip set debug on
SIP Debugging
On 8/20/14, 11:28 AM, Steve Totaro wrote:
PRI intense debug should show all you need to fix this.
Right, the sooner you post this debug here the sooner we can help.
Otherwise its just guesswork.
On Wed, Aug 20, 2014 at 12:13 PM, Jeff LaCoursiere j...@jeff.net
mailto:j...@jeff.net wrote:
On 8/11/14, 10:57 AM, David Wessell wrote:
I have an asterisk 1.8.x box that intermittently returns a 401. Calls
come through the same peer all the time, from the same carrier.
However intermittently the asterisk box returns a 401.
Below is the output of a failed call (1st) and a successful
On 8/7/14, 12:14 PM, D'Arcy J.M. Cain wrote:
On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad asghar...@gmail.com wrote:
Your call is up on VoiceMail you should check dialstatus before
sending user to VoiceMail.
I removed the voicemail command from the dialplan and it was exactly
the same
iptables -I INPUT 1 -p tcp --dport 5060 -m string --string VaxSIPUserAgent
--algo bm -j DROP
Its something like this
Registration from '30 sp:30@my_public_ip:5060 failed for
'192.168.xxx.xxx:6373' - Wrong Password
and there are approx 10 request per minute of this type.
Please suggest
What can I deduce from this? Is there some configuration on my
asterisk that can be tweaked so the failing requests can be handled
properly?
Is there additional information needed to make sense of this scenario
I suggest you capture the audio stream with tcpdump. You can then
convert the
On 4/7/14, 4:53 AM, Salaheddine Elharit wrote:
hello list,
i have a question i don't know if there is any possibility to stop
asterisk using a call for exp:
when i call a number 0522xx i want to excute a script or any idea
to stop asterisk automatically
Sure, try something like:
And I'm pretty sure if you look at any of those peers that have a
non-5060 port, the routers in front of them will rewrite packets
destined for ports 53277, 4121, 47822 etc. to the proper corresponding
internal IP:port where something is listening. The router of my
provider won't. It
On 2/6/14, 11:18 AM, Mike Diehl wrote:
Hi all,
I have an SPA112 that in sitting behind a Ubee cable modem. The
internet link is solid, but the device becomes unreachable within a
day or so of being rebooted. Then the customer goes to reboot the
device, they report that all 4 lights are
On 2/2/14, 9:42 AM, Markus Reschke wrote:
Hi!
My telco is Deutsche Telekom and they got about 30 SIP servers right
now. Currently I've set up a template for incoming calls in sip.conf
and added each SIP server by it's IP address like this:
[DTAG-in-1](DTAG-in-template)
host=217.0.16.103
On 1/28/14, 1:55 PM, motty cruz wrote:
Hi all,
I'm having issues with overwrite caller id, when I call someone my
caller id should be mycompanyinc but instead my id shows up as my
extension number 101.
this is what i have in sip.conf
[101]
type=friend
context=sipphones
call-limit=99
[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 TEI=0 Call Ref: len= 2 (reference 6918/0x1B06) (Sent
from originator)
[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 Message Type: RELEASE COMPLETE (90)
[Jan 14
David,
It seems to me that Asterisk is not seeing/binding to your VPN
interface. You need to debug that first. I would set en explicit bind
statement in sip.conf to the VPN interface address and nothing else.
Then start your asterisk and watch the log messages. It should confirm
that it
On 1/21/14, 4:38 PM, David Cunningham wrote:
Hi Andres,
Thanks for the idea. We did send bindaddr to the VPN address and
restarted Asterisk, but unfortunately that didn't solve the issue.
Asterisk didn't complain, but still the sip set debug on didn't show
the packets.
Ok, that is progress
On 1/16/14, 2:23 PM, Michael L. Young wrote:
- Original Message -
From: Andres and...@telesip.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 15, 2014 7:51:28 PM
Subject: Re: [asterisk-users] Asterisk ignoring nat
On 1/15/14, 6:11 PM, Leandro Dardini wrote:
Hello,
I have an asterisk box with a peer configured with
nat=force_rport,comedia, but asterisk keeps sending the audio to the
private IP address and ignoring the client peer nat settings.
Why don't you try with nat=yes. It should be equivalent to
On 12/31/13, 11:23 AM, Nick Olsen wrote:
Greetings all, First time poster, Sorry if this has been answered here
before.
We recently replaced a failed 1.4x asterisk PBX at a customer location.
Voicemail access was setup when the customer dialed *8, This worked in
1.4.
I suggest trying
On 12/18/13, 3:09 PM, alp...@gmail.com wrote:
Hello. I have a problem with the configuration of a remote extensions.
Calls are truncated at 20 seconds.
I got my my NAT firewall properly configured. Here I attached my debug
in CLI: http://pastebin.com/gh34E69f
When the call is setup I see
On 10/11/2013 10:05 PM, CDR wrote:
I am not proxying the media, but never the less I am forced to store
the source media IP in my CDR, for regulatory reasons. Asterisk gets
that information when the reinvite comes, but how do I store it?
If I don't figure this out my next email will be from
smime.p7s
Description: S/MIME Cryptographic Signature
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On 7/9/2012 8:24 AM, Sergio Serrano wrote:
Hi all
I hope that someone of you can solve this. Right now I'm stuck!
I'm using asterisk with some SIP extensions. Basically I want to
establish a call between desktop voip phone (ext 181) and embedded sip
system (ext 182)
All I can see in CLI
Considering that you made progress on your initial problem by setting
nat=force_rport (resulting in connected calls with no audio) and now
you're mentioning the use of externaddr, I'd recommend a very
careful reading of the NAT SUPPORT section of sip.conf.sample in the
configs directory of the
On 5/30/2012 2:34 PM, Eric Wieling wrote:
Has anyone experienced an issue with Sangoma analog cards where the card
suddenly stops working? Trying to dial out shows the channel as busy, even
though there is no active call on that port?
This happened to us often when we used Digium cards (in
peer is
being selected.
Andres
Besides, ulaw and alaw shows up when i do core show codecs audio in
the asterisk CLI, and there exists both codec_ulaw.so and
codec_alaw.so modules under the path /usr/lib/asterisk/modules/
I don't get it!...
More ideas?
Thanks,
Ricardo.
On Wed, May 9, 2012
No, as I understand an attended transfer, there is no 3-way period where the
receptionist introduces the caller to someone else. In an attended transfer,
from the caller's perspective, he's talking to the receptionist, then he's on
hold, then he's talking to someone else. No different from
-- Accepting call from '418nx2' to '418nx1' on channel 0/1,
span 1
-- Executing [418nx1@ael-default:1]
Answer(DAHDI/i1/418nx2-b, ) in new stack
-- Executing [418nx1@ael-default:2]
Wait(DAHDI/i1/418nx2-b, 2) in new stack
-- Executing
On 2/13/2012 10:49 AM, Andres wrote:
-- Accepting call from '418nx2' to '418nx1' on channel
0/1, span 1
-- Executing [418nx1@ael-default:1]
Answer(DAHDI/i1/418nx2-b, ) in new stack
-- Executing [418nx1@ael-default:2]
Wait(DAHDI/i1/418nx2-b, 2) in new stack
Error message on asterisk-1.4.39
chan_iax2.c:9541 socket_process: Rejected connect attempt from
192.168.141.8, requested/capability 0x2/0x703 incompatible with our
capability 0xc.
According to this log, server 192.168.141.8 has codecs defined as 0xc
(ulaw and alaw), which matches your
and maybe more but right now I don't recall any loopback device although I
won't be sure until I go to the site.
Can a loopback device be bought seperately?
Sure, we use the below device all the time:
http://www.amazon.com/Cables-Unlimited-TST-LOOP-003-Smartronix-Superlooper/dp/B000V6Y7IY
On 10/10/2011 10:31 PM, linux guy wrote:
On Mon, Oct 10, 2011 at 8:08 PM, Andresand...@telesip.net wrote:
I would recommend Acrobits. Not free but only a few bucks. It works fine
with ATT 3G.
This begs the question... which is more expensive (and where)...
making a regular cell
On 10/10/2011 5:35 PM, john Millican wrote:
Hello all,
Does anyone know of a good free/inexpensive 3G SIP client for the
iPhone? If anyone is using one that works good for them could you
please let me know.
Thank You,
JohnM
I would recommend Acrobits. Not free but only a few bucks. It
On 6/14/2011 5:08 AM, Paul Hayes wrote:
On 13/06/11 19:44, Mike Diehl wrote:
Hi all,
I'm trying to provision my PAP2T's to use a SVR lookup to find the
Asterisk
server. I'm using a provisioning file that contains an element like:
Proxy_1_ _sip._udp.example.com/Proxy_1_
However, the PAP
If you have experience with these phones...
We are trying to figure out how to transfer an established call on the
SPA504G while a second call is incoming. At present, the receptionist
has to answer every single incoming call before the XFER softkey is seen
again. This is completely
that it is the wrong variable name.
That makes no sense. You are configuring a PRI, not an analog line.
You should not be getting any messages regarding FXO or FXS. Take a
look at the file again and delete any reference to analog channels if
you are not using them.
Andres
http://www.neuroredes.com
I
hardware that
does not support 'hardhdlc', use 'dchan' instead.
Andres
http://www.neuroredes.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: August 30, 2010 11:45 AM
To: Asterisk Users
:
language=en
context=from-pstn
switchtype=national
signalling = pri_cpe
group=1
channel = 1-12
---
That is the most basic stuff you need to get the PRI up.
Andres
http://www.neuroredes.com
all 23.
Andres
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asterisk-users
to only send RFC2833.
Andres
http://www.telesip.net
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http
output all your variables to a file
and then you will learn if the variables do have the info you need.
Something like:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2',
etc /tmp/variables.txt;
system($message);
Andres
http://www.neuroredes.com
No success. Anybody please
On 7/26/2010 1:40 PM, Zarko Zivanovic wrote:
Hi Andres,
I did try what you said, but it didnt create any files:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2'
/tmp/variables.txt;
system($message);
This is what I do with Perl AGI scripts and it works fine. You
.
Andres
http://www.neuroredes.com
-- James
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the internet connection from your
server. Check and see if call processing works normally. If it
doesn't, do a tcpdump or ngrep capture to see what DNS queries are being
done and figure out why.
Andres
http://www.neuroredes.com
:
exten = s,n,Set(GROUP()=${custID})
GROUP in itself is not an application. Thats why you are getting No
application 'GROUP' for extension.
Andres
http://www.neuroredes.com
I'd like to count the number of simultaneous calls of one particular
customer (which may have several SIPaccounts
)
exten = _01XX,3,GotoIF($[${GROUP_COUNT(trunk1)} 4]?5)
exten = _01XX,4,Busy(1)
Andres
http://www.neuroredes.com
exten = s,n,NoOp(groepcount = GROUP_COUNT(${custID}))
The CLI shows :
[Jun 5 16:50:04] -- Executing [...@sub-settings:5]
Set(IAX2/testlocal-128, GROUP()=40
://downloads.asterisk.org/pub/telephony/sounds/
Andres
http://www.neuroredes.com
Hotmail: Free, trusted and rich email service. Get it now.
https://signup.live.com/signup.aspx?id=60969
show uptime
System uptime: 1 year, 17 weeks, 6 days, 17 hours, 57 minutes, 44 seconds
Last reload: 19 weeks, 8 hours, 12 minutes, 6 seconds
Andres
http://www.neuroredes.com
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other options in sip.conf and rtp.conf (relaxdtmf, directrtpsetup,
dtmftimeout) but none seem to make a difference.
Any help is greatly appreciated.
ANDRES--
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address in the Proxy field. The only way I could
recover the phone was to do a software upgrade. I think you don't even
have to change the software version.
--
Andres
Technical Support
http://www.neuroredes.com
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separate them with a comma.
Andres
http://www.neuroredes.com
disallow=all
allow=g729
to each of the extensions at the remote site. Then I did a SIP RELOAD.
So we'll see how that goes.
Thanks again for the assist - this has been quite an education.
Ben M. Schorr
Chief Executive Officer
the phone doesn't mean the other 5 lines are
registered. I bet one of those lines is the one with the actual odd
icon. It is probably a phone with a red cross. I have a SPA962 and
its exactly what shows under a specific line that cannot register.
Andres
http://www.neuroredes.com
Cheers,
j
your
default codecs there.
Andres
http://www.neuroredes.com
Thanks you,
Elliot
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switch does not like your CONNECT ACKNOWLEDGE message. I
have no idea why but my first guess would be to play around with the
'switchtype' in your chan_dahdi.conf. Another thing to try is to
enable/disable 'facilityenable' as well to see if it changes anything.
Andres
http
James A. Shigley wrote:
Never saw this appear on the list. So just resending it.
You might get more help if you include a PRI Debug that shows the call
being rejected.
Andres
http://www.neuroredes.com
Alright I’ve been having an issue when trying to dial out locally when
coming from SIP
if the simplex voice is corrected. If it is, you might
want to open a ticket with Digium since you have a problem with the
Hardware Echo Cancel module.
Andres
http://www.neuroredes.com
;
That's al I have for echo cancellation... I thought the hardware
module of Digium was of great quality
this happen with the aggressive echo cancel algorithms. You
might want to look into that.
Andres
http://www.neuroredes.com
Greetingz,
Jonas.
On Sat, 2009-05-30 at 14:35 -0400, Nathanial A. Byrnes wrote:
Hello,
I am working on a trixbox based system with a TDM410P connected to 3
phone lines
Hello
Please help me, i need transfer a call in asterisk to other telco number and
free the channel. Can i do with any q931 function?.
Thanks a lot
Aris...
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to materialize. I am currently
interested to know where it lives. Is it /etc/zapata.conf or
/etc/asterisk/zapata.conf?
It lives in /etc/asterisk/zapata.conf
You can find a sample file in the asterisk src 'configs' directory
called 'zapata.conf.sample'
Andres
http://www.neuroredes.com
Some
' = 'SPA2102'));
...and add the devices you need. As you can see we already added the
SPA2102.
Andres
http://www.neuroredes.com
Cheers,
j
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Jeff LaCoursiere wrote:
On Wed, 21 Jan 2009, Andres wrote:
Why don't you just edit the Trixbox endpoint manager files. They
produce basic XML files for Linksys and Polycom phones. It is trivial
to add support for any Linksys ATA as well.
File is:
/var/www/html/maint/modules
sent a DISCONNECT, your SIP phone sent a BYE, or your Asterisk
randomily hangup the call. Otherwise you are just guessing.
Andres
http://www.telesip.net
[snipped the beginning of this process...]
[Jan 9 12:34:12] VERBOSE[2778] logger.c: -- Executing [...@macro-dial:7]
Dial(Zap/9-1, SIP/2607SIP
= ACME. That way the
digest will come in as:
Digest username=ACME ...bla bla bla
Andres
http://www.telesip.net
I've done a SIP debug before, but I've done it again with the above
configuration:
No user '5551236049' in SIP users list
Found peer 'ACME' for '5551236049' from
-u100-usb-fxo-interface-device
http://wiki.sangoma.com/sangoma-wanpipe-usbfxo
Andres
http://www.telesip.net
Thank you.
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the correct
sip.conf entry to match the invite from the CS1500.
Andres
http://www.telesip.net
Frank
INVITE message from Wireshark packet capture:
INVITE sip:+15552027...@sip.acme.com SIP/2.0
From:
sip:5552022...@172.16.10.40;tag=f76c66d0-c7784528-13c4-2dbba4-767e6552-2db
ba4
To: sip:+15552027
Doug wrote:
Has anyone seen this before?
Yes, DNS problem.
Andres
http://www.telesip.net
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,
Andres,
http://www.telesip.net
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using
'ast_safe_sleep(chan,100);'.
Andres
http://www.telesip.net
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that with their regular landline or
cellphone.
Thanks,
Andres
http://www.telesip.net
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. The solution was to upgrade asterisk.
Andres
http://www.neuroredes.com
Gregory Malsack wrote:
Hey Everyone,
Here’s an email I received from a client who has a trixbox system that
has contracted with me for some custom dialplan programming
something with starting the driver from the command line.
If I start from boot or via a cron job, the value goes up to 99.99%.
I'm stumped.
Andres
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asterisk-users mailing list
need to me
made. http://wiki.sangoma.com/sangoma-wanpipe-smg-asterisk-bri-installation
Andres
http://www.neuroredes.com
Any help or suggestions would be gratefully appreciated :-)
Cheers
Phil
to the SERDEV mailing
list you would know that. The latest update is just from last week:
ser-2.0.1+cvs20081014_src.tar.gz 14-Oct-2008 06:26 2.5M
Andres.
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as I can remember. I am not sure about
version 1.6 though.
Andres
http://www.neuroredes.com
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asterisk
solution would be to enable the keep-alive
settings on the PAP2 and set it low to something like 15 seconds. The
setting is under the tab of line 1 and line 2 and its called NAT Keep
Alive Enable.
Andres
http://www.neuroredes.com
do you have qualify=yes ??
Is asterisk on a public IP
Josiah Bryan wrote:
Hey All -
Slight problem here - my install of 1.4.21.2 seems to be missing the
Queue application:
What does the CLI output say when you start asterisk and it gets to the
part where it tries to load app_queue.so?
Andres
http://www.neuroredes.com
asterisk*CLI core show
of the
times. My guess is your UA is either not doing SRV queries or is not
following them in any case. To know if it is actually doing the queries
try to sniff the traffic with Wireshark and analyze it.
Andres
http://www.neuroredes.com
thank you
regards,
nhadie
Andres
http
to your main server that it
cannot so it should try the next one in line according to your SRV Records.
We have deployed thousands of Linksys units configured to query SRV
records and they work fine in failover scenarios. I cannot comment on
X-Lite.
Andres
http://www.neuroredes.com
TIA
In other words, I'd really appreciate feedback from voip administrators (not
from resellers) who have had experience testing their devices and are happy
with them.
I would recommend the Linksys SPA8000 (8 port ATA). It is as solid and
reliable as the SPA2102.
Andres
http
will have No Access to the
parameter. The other 2 options are ro and rw, read-only and
read-write respectively. There is no way to define these directly on
the SPA web page, you have to use a provisioning file in order to define
these permissions.
Andres
http://www.neuroredes.com
Thanks,
Tom
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