Hoping someone can help me understand what is happening here;
we start asterisk as a service at boot (actually, with heartbeat) on
CentOS using the asterisk init script installed with make config
upon reboot of the server (when the asterisk service is first started by
heartbeat) we get color
.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Damon
Estep
Sent: Friday, September 18, 2009 9:28 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] console color
Hoping someone can help me understand what
: [asterisk-users] console color
On Friday 18 September 2009 09:28:24 Damon Estep wrote:
about once a month we issue a service asterisk restart via a cron
job,
and this is where we lose the color.
Most likely, your TERM environmental variable is not set when the cron
job
runs. This environmental
A new $500 USD bounty has been posted, should be a simple one!
http://www.voip-info.org/wiki/view/Asterisk+bounty+function+CURL+timeout
+v1.2+and+1.4
if this is too commercial for the lists liking my apologies in advance,
the resulting patch will be contributed to the project and will be
I have a need to disable the Send reply feature in asterisk voicemail
(1.2) because we have an environment where multiple servers use the same
real-time database for voicemail but the voicemail files are stored on
the individual server that the user registers with. When a user on
server A
Can anyone recall a bug where asterisk crashes after a callback agent
presses the ackcall key in 1.2?
The last logged item was before safe-asterisk restarted was delaying
member connect for 2 seconds
We have seen it two times on a heavily loaded server (1.2.12.1), but
cannot find anything
, to people that keep
these
types of issues in check by way of the official Digium bug database and
manage the process of fixing them.
On Wed, 19 Sep 2007, Damon Estep wrote:
Can anyone recall a bug where asterisk crashes after a callback agent
presses the ackcall key in 1.2?
The last
Has anyone designed a method to allow callback agents (Asterisk 1.2) to
log in on a Polycom SoundPoint IP phone and have the phone visually
indicate the agents logged in status?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
Answered my own question - use buddy watch on the Polycom and create a
hint priority extension for the agent channel...
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Wednesday, August 22, 2007 8:42 AM
To: Asterisk Users Mailing List - Non-Commercial
You can add the header the vendor is suggesting in asterisk as follows;
exten = #,1,SipAddHeader(P-Asserted-Identity:
sip:${CALLERID(num)[EMAIL PROTECTED])
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: Wednesday, August
Is there a clean way to disable music on hold for a specific user sip
user?
I have seen one example that creates a class called [none] that points
to an empty directory, which creates log errors that are annoying (but
not harmful?)
___
--Bandwidth
the Display IE?
Thanks and best regards to all,
Óscar Patrício
Anthony Francis wrote:
Damon Estep wrote:
Try putting a 1 second wait as step 1 in the dialplan, the SIP invite
is probably being send before the display IE arrives. The display IE is used
for CNAM delivery, and should
Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is
probably being send before the display IE arrives. The display IE is used for
CNAM delivery, and should not exceed 15 characters.
It is very common to put a message in the display IE that indicates that the
CNAM info will
There is a theory that says that jitter buffers should not be used until
the end of the voice path where jitter might be introduced. With that in
mind, and in this scenario, the jitter buffers should reside at the ATA
and media gateway;
ATA (SIP UA) ASTERISK NATIVE BRIDGE MEDIA GATEWAY (SIP
bridge
Damon Estep wrote:
Anyone know the answer? Has it been validated with packet captures,
or
code review?
All of the timing information should be passed across the bridge in
all of
the
frames that come in over RTP. I can't say I verified this with packet
captures,
but I did look
Same issue in 1.2.21 on FC6 with mpg123 0.66 (0.59r would not install on FC6).
What version of mpg123 are you using? What about Linux?
Here is my musiconhold.conf
[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3
there are mp3 files in the specific directory, and they are the default
There are many past posts stating that AMI is not stable when multiple
client computers are allowed to connect, particularly when connections
are dropped.
Has much progress been made on this? Is it more stable now than in the
past?
As of what versions were these issues improved?
Is it
It may not be exactly what you are looking for, but agentcallbacklogin
with ackcall=yes requires the # key to be pressed to answer a call from
a queue. We use this to avoid the possibility that the call ends up in a
cellular or home voicemail.
You can set the queued call to ring instead of
Damon Estep wrote:
http://www.asterisk.org/node/48317 does a nice job of explaining the
1.4 jitter buffer, however it raised a question in my mind.
In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the
UDP RTP packets renumbered on transmit, or is the original
http://www.asterisk.org/node/48317 does a nice job of explaining the 1.4
jitter buffer, however it raised a question in my mind.
In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the UDP
RTP packets renumbered on transmit, or is the original sequence number
preserved in the UDP
There are 6 different ${QUEUESTATUS} variable values defined in asterisk
1.2, I am attempting to make sure I have a full understanting of when
they would be set;
If someone could correct errors with these definitions ot would be
appreciated;
TIMEOUT - the max time specified in the queue
Who is tired of dealing with DST changes?
I have asterisk running on FC4, FC4 has been patched and shows the
correct MDT timezone and time.
Email notifications of voicemail show the message time an hour early
(standard time, not daylight). This si the time in the message body, not
the
, Damon Estep wrote:
Who is tired of dealing with DST changes?
I have asterisk running on FC4, FC4 has been patched and shows the
correct MDT timezone and time.
Email notifications of voicemail show the message time an hour early
(standard time, not daylight). This si the time
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roberto
Sent: Tuesday, February 27, 2007 7:28 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] RES: asterisk-users Digest, Vol 31, Issue 115
Questions:
Does anyone have a really STABLE
Why do that?
Just traffic shape each user/group of IP addresses to the total
bandwidth you want them to have and then set up a low latency queue for
voip traffic, that way the voip bandwidth can be used for data when
there are no calls but will give VoIP traffic priority over other
traffic.
Any
Are all of the sip phones in the same context?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Robert Jenkins
Sent: Tuesday, January 16, 2007 1:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
You are trying to subscribe to a non SIP channel?
Not sure that can be done...never tried.
-Original Message-
From: Robert Jenkins [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 16, 2007 4:43 PM
To: Damon Estep; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE
] On Behalf Of Damon
Estep
Sent: Thursday, January 04, 2007 1:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk sip peer/user matching methods
forauthentication backwards?
Take an example where there is two sip users defined in sip.conf
In order to work around some authentication issues I am considering
connecting two asterisk boxes with IAX instead of SIP. The original
reason for choosing SIP was to reduce the need to translate SIP
signaling to IAX, since all origination, termination, and UAs are SIP.
Can anyone comment on
I have native music on hold setup to play ulaw encoded files. No
transcoding, caller is on a g.711u SIP channel. There is horrible
distortion and noise between files for 1 to 2 seconds.
Has anyone seen this? I check the files and trimmed silence from the
end, the source of the noise is not the
Take an example where there is two sip users defined in sip.conf as
follows;
[peer1]
Host=192.168.1.1
...
[peer2]
Host=dynamic
Secret=password
...
[Peer3]
Config not relevant
...
The intention is to accept calls from peer1 without authentication (ip
address authentication
:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip peer name channel variable?
Check out this page:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sipchaninfo
bp
On 12/17/06, Damon Estep [EMAIL PROTECTED] wrote:
Started out
Started out looking for what I thought was going to be a simple variable name,
have not found it.
Does anyone know of a variable that would contain only the SIP peer name of the
originating channel?
${CHANNEL} contains it, but it needs to be parsed and our dial plan sometimes
uses local
will be paid when (and if) the patch is merged into to
code tree for the 1.2 branch, which will obviously require favorable
peer review.
If this issue appears larger than a couple of hours work I welcome the
feedback.
Damon Estep
[EMAIL PROTECTED
Your dial string must have either the t or T option set.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of omar
parihuana
Sent: Friday, December 01, 2006 9:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users]
Caller ID should always be either ANI + CNAM (where available) on
inbound, or anonymous (No ANI).
If you are getting anything different from your Telco something is
wrong.
For SIP originated calls the CID is derived from the INVITE
Outbound caller ID is as you set it in your peer/user config.
] On Behalf Of omar
parihuana
Sent: Friday, December 01, 2006 5:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CALL TRANSFER
Thanks!!!
I forget Tt option! (too basis!!)
On 12/1/06, Damon Estep [EMAIL PROTECTED] wrote:
Your dial string must
I have seen a couple of posts related to this, but no workaround.
Setup;
Asterisk 1.2.13 with Polycom IP501 phones
Caller is sent to the queue with the t option
Agent is logged in via AgentCallbackLogin on an extension that is in a
context that includes exclusively agent extensions.
Can anyone suggest a reason why these channels might end up zombies?
The process is;
Call comes in via SIP into a context that appends the caller ID name as
follows;
[cnam-lookup]
exten =
_[2-9]X,1,set(CALLERID(name)=${CURL(http://cnam.provider.com/?co
That is a huge question, but the short answer is;
They sent you s SIP invite to the [EMAIL PROTECTED] including
whatever credentials are required to authenticate them based on how you
have them defined in your sip.conf.
You could allow anonymous, but be careful that the context it comes into
Which version?
Similar issues parsing callback number in 1.2.12
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner
Sent: Thursday, September 28, 2006 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial
Can anyone tell me if accountcode= should appear before or after the
channels definition that you want it to apply to?
I have several groups of channels (PRI) defined in Zapata.conf, and wish
to specify accountcode by group.
If I put the accountcode= at the end of Zapata.conf it applies to
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] wget from within asterisk?
You need curl-devel just try
yum install curl-devel
Damon Estep wrote:
On version 1.2.12.1 running on FC4 with curl.i386 installed the asterisk
CURL function is not registered
Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] wget from within asterisk?
The Curl/CURL is an asterisk dialplan distinction.
-Original Message-
From: Damon Estep [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
What would be the simplest way to retrieve information form a CNAM
database that provides http based query responses?
Does an application or script already exist that does this?
Basically, I want to do a wget of a URL that contains the callerID
number as a variable, and assign the returned
-Commercial Discussion
Subject: Re: [asterisk-users] wget from within asterisk?
-Original Message-
From: Damon Estep [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Fri, 17 Nov 2006 13:32:53 -0700
Subject: [asterisk-users
: [asterisk-users] wget from within asterisk?
Make sure the curl library/package is installed, then re-compile
asterisk. We're using it on 1.2.
-Original Message-
From: Damon Estep [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
- Non-Commercial Discussion
Subject: RE: [asterisk-users] wget from within asterisk?
They both seem to work, but the Curl spits out warnings about being
deprecated. Ours are all configured using CURL.
-Original Message-
From: Damon Estep [EMAIL PROTECTED
] On Behalf Of Damon
Estep
Sent: Friday, November 17, 2006 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] wget from within asterisk?
Thanks a bunch, this seems to be a simple solution, I just did not have
CURL installed before I built asterisk
: Damon Estep [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Fri, 17 Nov 2006 13:40:40 -0700
Subject: RE: [asterisk-users] wget from within asterisk?
I saw CURL, but it does not register appear in show
Most usage charges are stored in various
billing databases as per MINUTE of use, not per 6 seconds of use.
6 second billing simply means that you
bill in decimal fractions of a minute, 66 seconds becomes 1.1 minutes.
1. Divide your billsec value by 60 and
round to 1 decimal place. Add
Is there an option in meetme.conf or the application meetme
to set a strict participant count limit on a per room basis?
I checked the sample meetme.conf and did a show application
meetme, as well as a couple of Google searches and came up empty handed.
This is for a system with SIP
Keep in mind that CDR records show calls
sent to VM as answered, so you also have to look at the lastapp field
Disposition=answered and lastapp=voicemail
means the call was answered by voicemail (obviously)
If you are doing billing you do not care, because
the are both billable, but
I have 2 asterisk boxes connected via SIP
box 1 sip peer connected
to box 2 (ip addresses intentionally removed)
[ast20]
type=friend
host=x.x.x.20
insecure=very
context=subscriber
dtmfmode=inband
qualify=no
canreinvite=no
disallow=all
allow=ulaw
box 2 sip peer connected
I have several diaplans that used the exten a
to handle the * key. They all stopped working some time in the
last few releases of *
Was a intentionally dropped as a standard extension
in favor of using * in the dialplan?
Example;
Exten = a,1,dosomething used to work, now you must
Can anyone suggest any reasons why a zap (PRI) b channel
should not be a member of multiple zap trunk group definitions?
For example;
Group 1 = Channels 1 to 23
Group 2 = channels 1 to 12
Group 3 = channels 13 to 23
The purpose is to restrict the number of channels a
particular
Damon Estep wrote:
Can anyone suggest any reasons why a zap (PRI) b channel should not
be a
member of multiple zap trunk group definitions?
For example;
Group 1 = Channels 1 to 23
Group 2 = channels 1 to 12
Group 3 = channels 13 to 23
The purpose
Secure multi-tenant partitioning capabilities?
What is your distribution intentions, commercial or GPL?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Sunday, October 15, 2006 10:33 PM
To: Asterisk Users Mailing List
Anyone encountered this on yet?
WARNING[23251]: chan_sip.c:2570 sip_write: Asked to transmit
frame type 64, while native formats is 4 (read/write = 4/4)
Started after an upgrade from CVS 8/2005 to current 1.2.12.1
If I had a reference for what frame types 4 and 64 are I might
be
Try taking to 90 second timeout off
Change
exten = _[1-9].,4,Dial(SIP/${enumresult},90)
to
exten = _[1-9].,4,Dial(SIP/${enumresult})
a btter method is to set up each office as a unique peer with qualify = yes and
then add the peer name to the dial string, like dial(SIP/[EMAIL PROTECTED])
if
I had read a post somewhere that there is an XML parameter for
the Polycom config files for default handset volume, but I can not locate it
again.
Anyone know what it is?
I want to set the default handset volume higher on some
phones, despite the ADA hearing aid warning in the admin
I have the hint priority defined for a few SIP
phones.
When I make a call OUT from one of the phones I see that the
show hints picks up a status change from 0 to 1 for the extension,
but when I call IN to that extension the hint status is still 0.
This is on a server built back in
According to the wiki page http://www.voip-info.org/wiki/view/Asterisk+IAXmodem
There are a couple of ways to integrate Asterisk and HylaFax
with IAXmodem;
IAXmodem as HylaFax modem,
both HylaFax and Asterisk on the same machine
IAXmodem in conjunction with
Do you have audio running during the hold (MOH), or silence?
Could the Polycom (or asterisk) be dropping the call due to inactivity?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak
Sent: Friday, August 11, 2006 6:04 AM
I have seen a very small number of posts on this type of
setup;
1. mysql replicated failover cluster (Linux HA) for the
realtime databases and ODBC voicemail storage
2. multiple asterisk servers (~4) connected to the SAME
realtime tables and VM store.
3. Any defined SIP client
Here is the config for one of several boxes we run in similar
environments;
A dell SC1425 1u rackmount with dual Xeon CPUs, 1GB ram, dual 80gb sata
drives (software raid 1), fedora core 4, and a sangoma a104 4 port T1
interface card.
A good choice for business quality SIP phones is the Polycom
This was a topic covered a day or so ago.
I asked this same question, and my Cisco Voice product rep explained
that the 02 series has more memory to handle larger firmware images.
The two models take different firmware, and some newer features will not
be able to be implemented on the non 02
Search the wiki for the application command realtime() if you are using
realtime.
www.voip-info.org
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of unplug
Sent: Monday, June 12, 2006 10:47 AM
To: Asterisk Users Mailing List -
Turn off echo can for those calls.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Monday, June 12, 2006 10:55
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] TDM Fax
Problems
I am running into
The most significant change in the 02 versions of the SPA line is more
memory to handle larger firmware images.
They do not use the same firmware as the non 02 models and will reject
the older images.
As the firmware image evolves it gets larger, and the previous model
will end up being limited
.
exten=101,1,Answer()
exten=101,2,NoOp(${CANCALLFORWARD})
How cant I get the value of each field in the table?
On 6/13/06, Damon Estep [EMAIL PROTECTED] wrote:
Search the wiki for the application command realtime() if you are
using
realtime.
www.voip-info.org
from DB directly
Thanks!
Do you mean there is a realtime function available to get and set the
value in table? Can you give me some references (website) as I have
found nothing of this function.
On 6/13/06, Damon Estep [EMAIL PROTECTED] wrote:
The question has changed, but the answer has
Sox will do it, the syntax is a little tricky and I am not an expert
with sox.
Also, check to see if you are using quietmp3 or the current equivalent
in your moh config file.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michael
There is not a formula, but I second the opinion that the config is
adequate if the linux build behaves on the hardware (correct drivers,
config, etc).
The limiting factor is the E1, you will be able to handle a full E1 of
traffic, with transcoding, with this box.
This is not based on a formula,
Not trying to be rude, but you will either need to invest many, many
hours learning how asterisk works and evaluating 3rd party billing
solutions, or possibly writing your own.
This will require light programming skills (agi, mysql, perl, etc), but
probably not C unless you really want to
What you are proposing is quiet simple, and is done regularly. We
provision Linksys Sipura ATAs via a perl script with SSL and client
certificate authentication, as well as Polycom phones via XML file
drops. The newest Polycom firmware also states that ssl is supported,
but we have not made the
Here is an example of a dialplan that
looks up a post dial code in a mysql database and updates the accountcode
accordingly.
exten =
_1XX,1,Read(postcode|beep|3||1|10)
exten =
_1XX,2,set(level_auth=0)
exten =
Is anyone aware of a method to signal back to a caller that
the party they have called is on another call (other than the obvious -send
to VM or a busy signal).
What I am thinking of is the alternate ring you get when you
call someone on various cellular networks or PSTN networks where
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mike Fedyk
Sent: Friday, June 02, 2006 10:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom-Asterisk hints/presence
How do you
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mike Fedyk
Sent: Thursday, June 01, 2006 11:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] addons trunk make error
There are too
I set up hints and presence monitoring on some Polycom phones
connected to an asterisk server with the expectation that the phones that are watching
other extensions would be notified when the other extension sis ringing, in
addition to the other statuses (on the phone, statuses set by the
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob Amen
Sent: Thursday, June 01, 2006 5:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom-Asterisk hints/presence
Damon Estep wrote:
My goal was to have someone's assistant see that the boss's line
On 6/1/06, Damon Estep [EMAIL PROTECTED] wrote:
Thanks Bob,
We have also done some similar stuff to make it usable, the prospect
that we might be able to achieve the same functionality and add to
it as
a bonus the ability to monitor the boss's extension state (idle, on
the
phone
You're right in that there is nothing in technology spec to support
the concept of shared line appearance, but I think what was more to my
point was that you could get access to a shared line from more
channels than just a SIP channel. I'd probably want the ability to
have two SIP
From Kevin Flemming at Digium:
It will most definitely not be in 1.4, but I
would expect it to appear some time early in the next development
cycle
and be part of Asterisk 1.6.
Sean,
Where did you find that quote, I would like to see the rest of the
thread if there was relevant
Changing the subject back to the original topic;
Is there bug in the asterisk hint/presence implementation, or an
intentional omission, or a lack of understanding on my part?
A SIP debug of a subscribed extension shows that asterisk only sends the
SIP presence notification to the subscriber when
I think there was a patch that went in recently from Mark with regard
to SIP devices and their state when they are ringing/in use and when
they are just in use. That may help you with what you're asking
about.
I tested on 1.2.8, was it after the release of 1.2.8?
If your comments echo those of past conversations on the matter I
can
see that a bounty at this point would not be money well spent, since
any
work the comes from it is not likely to make the cut. A bounty would
only be useful to accelerate the implementation of a feature where
there
I think there was a patch that went in recently from Mark with regard
to SIP devices and their state when they are ringing/in use and when
they are just in use. That may help you with what you're asking
about.
Let's assume for a minute that there is a way to get a ringing notify,
and
@suburbanbroadband.net wrote:
I think there was a patch that went in recently from Mark with
regard
to SIP devices and their state when they are ringing/in use and
when
they are just in use. That may help you with what you're asking
about.
I tested on 1.2.8, was it after
Anyone run a make on asterisk-addons /trunk r219 ?
I error out on mp3 on a FC4 box, and I do not see anything
obvious (to me) in the errors.
make[1]: Entering directory
`/usr/src/addons-trunk/format_mp3'
gcc -pipe -fPIC -Wall -Wstrict-prototypes
-Wmissing-prototypes
for you?
-Jason
Damon Estep wrote:
Can someone shed some light on why the 'hint' feature was
implemented
in the 'priority' field that is purely an integer in the rest of the
dialplan?
There seems to be a conflict with realtime and the hint priority, in
order to put in the hints you
I have a production server running a CVS Head release dated
8/27, which is pretty much 1.2 minus some last minute additions, 1.2 was
released at the end of august 2005.
There is a sip channel patch related to presence and sip
subscriptions that I wish to apply, but since the server has
Can someone shed some light on why the hint feature
was implemented in the priority field that is purely an integer
in the rest of the dialplan?
There seems to be a conflict with realtime and the hint
priority, in order to put in the hints you would have to change the priority
column in
. Asterisk
wasn't
designed with multiple companies in mind.
-Original Message-
From: Damon Estep [mailto:[EMAIL PROTECTED]
Sent: Wednesday, May 17, 2006 11:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Providers using Embedded
A few things;
You have nat and qualify = yes, those settings are correct.
On your DSL, is there a public IP address on the internet side of the
Linksys? (not in the 10.x.x.x, 192.168.x.x, or 172.16.x.x subnets).
If not, you have another NAT router in the middle (your DSL modem) and
you will not
I get the impression the complaint is NO audio, not poor audio. This
points more to NAT than QoS.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek
Lee-Wo
Sent: Wednesday, May 17, 2006 1:38 PM
To: Asterisk Users Mailing List - Non-Commercial
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Strom Carlson
Sent: Wednesday, May 17, 2006 11:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Plan to free myself from AAH
On
Most of the VoIP service providers I have encountered are moving in a
different direction, with a goal of NOT having any customer premise
equipment other than the SIP hard phones, soft phones, and ATAs, along
with an IP access router with QoS.
-Original Message-
From: [EMAIL PROTECTED]
CHANNEL variables are lost when the CHANNEL is hung up and torn down, so
while you can pass the variable down by setting it with the double
underscore, it will not be set when you start at the top again.
There might be other ways to accomplish your goal, one would be to use a
global variable that
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