On Fri, 2019-10-11 at 14:12 -0400, Brian J. Murrell wrote:
> I'm trying to clarify my understand of gosub, macros and AEL.
>
> My understanding is that macros using the Macro() application, which
> is
> defined in extensions.conf by:
>
> [macro-foo]
> ...
>
>
I'm trying to clarify my understand of gosub, macros and AEL.
My understanding is that macros using the Macro() application, which is
defined in extensions.conf by:
[macro-foo]
...
and called in extensions.conf with
exten => _9NXXNXX.,n,Macro(fastbusy)
is deprecated in favour of Gosub().
Using Asteirsk 13.28.1:
If I configure my pjsip transport to handle NAT from the Internet:
[transport-tcp]
type=transport
protocol=tcp
bind=10.75.22.8:5060
local_net=10.75.22.0/24
external_media_address=[external address redacted]
external_signaling_address=[external address redacted]
When a cal
On Fri, 2019-09-13 at 14:21 +0200, Administrator TOOTAI wrote:
>
> Escape it with \
Tried that. It doesn't work.
Cheers,
b.
signature.asc
Description: This is a digitally signed message part
--
_
-- Bandwidth and Colocation
How can I use an IF statement with a true value being a variable that
has a colon in it? The colon in the true value variable is being taken
as the delimiter for the false value.
The only solution I came up with was some hackery to use STRREPLACE to
replace the : with a % before the IF statement
On Tue, 2019-07-16 at 16:20 -0400, Brian J. Murrell wrote:
> Is there any option to prevent Asterisk from rewriting the To:
> address
> of a SIP MESSAGE that it's received and will forward to another SIP
> client?
>
> That is, when Asterisk receives a MESSAGE with the To;
Is there any option to prevent Asterisk from rewriting the To: address
of a SIP MESSAGE that it's received and will forward to another SIP
client?
That is, when Asterisk receives a MESSAGE with the To; header saying:
To:
and wants to forward that to foo@10.75.22.100, I'd like the To: header
to
On Thu, 2019-06-06 at 09:33 -0400, Brian J. Murrell wrote:
> I'm trying to use linphone-android with asterisk but there is an
> aspect
> of the way asterisk and linphone-android interact with MESSAGE
> transactions that is causing problems.
>
> The linphone-android folks co
I'm trying to use linphone-android with asterisk but there is an aspect
of the way asterisk and linphone-android interact with MESSAGE
transactions that is causing problems.
The linphone-android folks consider both the To: and From: address in
MESSAGE transactions when deciding which "chat" to put
On Wed, 2019-04-17 at 13:50 -0400, Joshua C. Colp wrote:
>
> The same escaping should apply there for extensions.conf as it's a
> config file thing, I don't use AEL and don't know anything in that
> regard. It may work the same way there.
How very odd. It is working now. I am sure I did nothing
On Wed, 2019-04-17 at 11:56 -0400, Joshua C. Colp wrote:
> On Wed, Apr 17, 2019, at 12:51 PM, Brian J. Murrell wrote:
> >
> > I can add it onto the end of the variable in the Dial() command:
> >
> > Dial(${FRED};transport=tcp,${timeout},TtWw);
[ the part you trimmed
On Wed, 2019-04-17 at 10:04 -0400, Joshua C. Colp wrote:
>
> You specify the transport in the SIP URI. For example:
>
> sip:t...@example.com;transport=tcp
Hrm. This is probably going to be pretty basic, but some googling
didn't seem to come up with anything. How do you do this when you are
ass
Hi,
I'm using Asterisk 13.x and have defined a pjsip TCP IPv6 transport:
[transport-tcp-ipv6]
type=transport
protocol=tcp
bind=[2001:1234:5678:abcd::2]:5060
I also have an IPv4 version of that:
[transport-tcp-ipv4]
type=transport
protocol=tcp
bind=10.75.22.8:5060
I've then configured an endpoi
On Thu, 2019-04-04 at 15:08 +0200, Antony Stone wrote:
>
> It's not "Password", it's "Secret" :)
Ha ha. I knew it would be a head-smack type problem.
Cheers,
b.
signature.asc
Description: This is a digitally signed message part
--
I'm not sure how much more simple I can make this but I just cannot
seem to get my Asterisk 13 to accept a connection on the manager
interface:
--- manager.conf ---
[general]
enabled = yes
port = 5038
bindaddr = 127.0.0.1
[myasterisk]
secret=a
permit=0.0.0.0/0.0.0.0
read = all
write = all
So,
On Thu, 2019-02-21 at 11:17 -0500, Brian J. Murrell wrote:
> In the past, I have created variables that hold multiple extensions
> such as:
>
> HOUSEPHONES=PJSIP/mom&PJSIP&dad&PJSIP/grandma
>
> so that I can do a Dial(${HOUSEPHONES},...) with it, to ring multiple
On Fri, 2019-03-01 at 15:54 -0500, Joshua C. Colp wrote:
>
> That's correct. You'd either need to retrieve the line parameter from
> the outbound registration or forge the source IP address,
Can I eliminate the identify by IP address then, given that my ITSP is
supporting the line parameter? Or
On Fri, 2019-03-01 at 15:41 -0500, Joshua C. Colp wrote:
>
> I don't understand what you mean. Your ITSP has stated that they
> don't want you to do authentication with them, so you can't.
They are implying, as I am understanding them, that somehow SIP packets
they send me shouldn't need to be au
On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote:
> you can try line functionality on the outbound registration which
> may or may not work[2] (requires the upstream to adhere to the RFC,
> which not all do).
My provider seems to implement this.
However even with the line=... in the:
SIP
On Fri, 2019-03-01 at 14:15 -0500, Joshua C. Colp wrote:
>
> You either configure IP based matching using an identify section[1]
That's what I did:
[itsp]
type=registration
transport=transport-udp
outbound_auth=itsp-auth
server_uri=sip:pop1.itsp.example.com
client_uri=sip:x...@pop1.itsp.example
I'm being told by my ITSP that my Asterisk shouldn't be challenging
their system to authenticate (i.e. a 401 response) when they send me a
SIP MESSAGE (or I suppose a SIP INVITE for that matter).
But I'm not sure what a pjsip.conf configuration for that looks like.
How does one associate an incom
On Sun, 2019-02-17 at 17:31 -0500, Brian J. Murrell wrote:
> I have a PJSIP trunk set up which works fine for voice. I can call
> out
> and I receive calls from it once it registers.
>
> What isn't working though is receiving MESSAGE (i.e. SIP SIMPLE)
> events. It was wo
In the past, I have created variables that hold multiple extensions
such as:
HOUSEPHONES=PJSIP/mom&PJSIP&dad&PJSIP/grandma
so that I can do a Dial(${HOUSEPHONES},...) with it, to ring multiple
phones.
But now some of those phones will be registering multiple times and
thus have multiple contacts
On Wed, 2019-02-20 at 12:38 -0700, John Kiniston wrote:
> I don't see any other messages in this thread other than your initial
> one
> and my response, perhaps the listserv hasn't relayed it to me yet.
I started a new thread:
http://lists.digium.com/pipermail/asterisk-users/2019-February/293668.
On Wed, 2019-02-20 at 11:46 -0700, John Kiniston wrote:
> Use the IF function to evaluate and change the dial command directly.
Thanks for taking the time, but that doesn't actually answer the
question I asked. It in fact answers the caveat I specifically
mentioned:
> Granted the particular abov
Following up on my previously asked question if I rewrite the branching
example (not that it negates the more general branching question) I was
using as such:
exten =>
s,n,Set(EXT=${IF($[${SIP}=PJSIP]?${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,PJSIP/,)})}:${ARG2})})
exten => s,n,Dial(${EXT},20,TtWw)
Is there any less cumbersome way of doing conditionalized/branching in
extensions.conf other than something like:
exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip)
exten => s,n,Dial(${ARG2},20,TtWw)
exten => s,n,Goto(afterdial)
exten =>
s,n(pjsip),Dial(${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJS
I have a PJSIP trunk set up which works fine for voice. I can call out
and I receive calls from it once it registers.
What isn't working though is receiving MESSAGE (i.e. SIP SIMPLE)
events. It was working earlier today but I seem to have done something
as I was enabling voice on the trunk to me
On Mon, 2019-01-28 at 07:29 -0700, George Joseph wrote:
> When you have an "identify" object configured, you should just use
> "ip" as
> the "identify_by",
But isn't ip the highest priory check in the default value of
endpoint_identifier_order and by extension, wouldn't an endpoint
without an "ide
On Mon, 2019-01-28 at 07:29 -0700, George Joseph wrote:
>
> What version of Asterisk
13.11.1
I know, I could stand to upgrade.
> and what's the value of the "identify_by"
> parameter for the endpoint?
It doesn't have one. I guess you are implying it should have one.
> When you have an "identi
I have a trunk set up for the DID from my provider:
[my_provider]
type=registration
outbound_auth=my_provider
server_uri=sip:sip.example.com
client_uri=sip:my_usern...@sip.example.com
retry_interval=60
[my_provider]
type=auth
auth_type=userpass
password=123456
username=my_username
[my_provider
On Tue, 2019-01-15 at 12:01 -0500, Joshua C. Colp wrote:
>
> The chan_sip module has this implemented under the "nat" option using
> "comedia" as I recall.
Yeah. The help for which reads:
Send media to the port Asterisk received it from regardless
of where the SDP says to send it.
> It causes
On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote:
> How is your endpoint currently configured in asterisk?
It's configured as a chan_sip peer.
> Have you tried
> rtp_symmetric to see if the endpoint sends audio to asterisk if
> asterisk
> can send audio back to the client?
That would requ
This is going to be a bit of an odd situation, but perhaps might become
more and more common (as mobile phone SIP clients utilize PUSH proxies
instead of the battery draining direct registering with SIP servers).
I have a SIP client which can be on the same RFC-1918 LAN as my
Asterisk server. Eve
Hi,
I want to be able to send SIP SIMPLE messages via/to my VOIP provider
but in trying to do so with MessageSend() I am getting 401 errors back
from them, unsurprisingly. They want such messages from me
authenticated with my account just as they would for SIP voice calls.
For voice calls, of co
Running 15.1.2. I have four devices transitioned to use pjsip.
After about 1-2 days of uptime, psjip stops accepting registrations,
and the messages log contains the entry as per the subject.
At any given time, "pjsip show contacts" only shows the four devices.
Could someone point me to a fix,
I'm running Asterisk 15.1.0 and in the process of converting my
various SIP endpoints to use PJSIP.
My Zoiper client causes the messages quoted below to show up on the
CLI once per minute. Things seem to work OK, but I am curious because
there seems to be no way to suppress the messages, and ther
On Sat, Jul 30, 2016 at 7:18 AM, D'Arcy J.M. Cain wrote:
>
> Bad, bad idea. If you remove the password then anyone can get to the
> mailbox.
Depends on your use case, at home I have several phones and one mailbox. So
I _want_ everyone to get to the mailbox with a minimum of
x
with no auth, not just removing the prompts. Try something like this
exten => *85,1,VoicemailMain(100,s)
When you dial *85, you will get voicemail for mailbox 100, and no passcode
prompt. Jumps straight into the menu "You have FOUR new messages..."
--
Brian Wilson
Wildsong
--
___
one
or some other sound in it as feedback.
The file has to be in the right format for asterisk, after recording it on
a Windows machine, I used this command to convert it:
sox infile.wav -r 8000 -e signed-integer -b 16 -c 1 outfile.wav
"Sox" figu
Reviving an old thread, still seeing this.
Brian Wilson wrote:
>* I've been getting slammed with these messages on my console lately.
*>>* ed -1: Invalid argument
*>* [2016-05-31 10:09:40] WARNING[16249]: chan_sip.c:3775 __sip_xmit:
*>* sip_xmit of 0x7f05140803f0 (len 559) to
-- adding another media channel for video should not affect
audio.
I believe I have UDP ports 5000-4 open right now on the firewall.
I also don't understand why it varies from day to day.
Any ideas on how to debug or what might be happening?
--
Brian Wilson, GISP
Wildsong: 707-827
llo
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Brian Wilson, GISP
Wildsong 707-827-0001
--
_
-- Bandwidth and Colocation
--
I am guessing this is some apparently unrelated and undocumented side
effect of a setting I have changed recently but have not been successful
chasing it down so far. I have this happening on two different servers now.
--
Brian Wilson
Wildsong 707-827-0001
--
___
-r" immediately after starting asterisk and watch
for error messages. Be warned that sometimes the errors will lead you far
far astray. Usually they are useful.
Brian
--
_
-- Bandwidth and Colocation Provided by http://www.
codecs (lpc10 and ilbc) because
> they used an instruction that doesn't exist on the server (it's an oldish
> HP mini-server). I'm guessing from the above message that VM might be
> afflicted by the same issue. Presumably compiling from source will solve
> this? (I've c
cruft left over
from previous builds or installations. Things like this are more likely to
show up that way.
Brian
On Thu, May 5, 2016 at 10:21 AM, Michael Ströder
wrote:
> Joshua Colp wrote:
> > Michael Ströder wrote:
> >> Joshua Colp wrote:
> >>> Michael Ströder wro
Clever. Some packages are missing though. Should I convey this to someone?
On Thu, Mar 31, 2016 at 12:28 PM, George Joseph wrote:
>
> Run ./contrib/scripts/install_prereq. I think your'e missing the
> python-dev package. I'll update the Wiki.
>
>
--
_
The way I got this build to succeed last night was by using a separate
pjproject, error I get with bundle is the same after applying your patches.
First patch succeeds.
Second patch fails in 'configure'.
What I did -- I downloaded your diffs, unpacked a fresh copy of the
asterisk tarball, then ap
gt; https://gerrit.asterisk.org//2516
> https://gerrit.asterisk.org/2449
>
> Any other feedback? I'd like to get an idea of how many folks have tried
> it.
>
> --
Brian Wilson, GISP
Wildsong 707-827-0001
--
__
:39 AM, Marek Červenka wrote:
> Dne 28.1.2016 v 13:37 Brian :: napsal(a):
>
> when you say load - how many concurrent calls? Is there transcoding
> happening? sip / PRIs ? what load?
>
>
> 12 concurrent calls
>
> no transcoding
>
> SIP
>
> under 1.5 with 4x 1Ghz
when you say load - how many concurrent calls? Is there transcoding
happening? sip / PRIs ? what load?
On Thu, Jan 28, 2016 at 9:57 AM, Marek Červenka wrote:
> Dne 27.1.2016 v 17:50 A J Stiles napsal(a):
>
>> On Wednesday 27 Jan 2016, Marek Červenka wrote:
>>
>>> Dne 27.1.2016 v 13:14 A J Stiles
sip trace?
On Mon, Dec 21, 2015 at 6:56 PM, Luca Bertoncello
wrote:
> Karsten Wemheuer schrieb:
>
> Hi Karsten!
>
> > the timeout value of 15 minutes directs me to an issue with session
> > timer. Try to refuse them by putting the line
> > session-timers = refuse
> > into the general co
There has been some real stupid stuff going on in the inter carrier market
recently.
One of them was to attach a massive premium €2.50 per minute to calls to
Switzerland sunrise mobile for example.. This was only if your CLI was of
certain countries or invalid.
I don't know anything about voip.ms
add a pause in the dialplan for a second then proceed..
On Wed, Nov 25, 2015 at 2:27 PM, Tony Mountifield
wrote:
> In article <20151125133008.6369360.14455.17...@gmail.com>,
> Israel Gottlieb wrote:
> > Try putting progress instead of answer
>
> Yes, I tried Progress already, and it didn't he
probably opensips isn't forwarding the re-invite to the endpoint.. set
re-invites up and run sip tracing on your opensips and asterisk box and see
what happens when the reinvites arrive.
On Sat, Nov 21, 2015 at 8:10 PM, Steve Edwards
wrote:
> On 11/20/15 11:13 AM, Steve Edwards wrote:
>>
>
> I h
s well:
http://www.amazon.com/TRENDnet-8-Port-100Mbps-Switch-TPE-S44/dp/B000QYEN
1W/ref=sr_1_10?ie=UTF8&qid=1426296706&sr=8-10&keywords=poe+8-port
Brian Franklin
NTG, Inc. - "Problem Solved"
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-b
t every time the system boots Then how to configure
> the channels " chan_allogsm" ..
>
You'll need to install the basics like kernel headers and build
essentials to compile any modules for your running kernel. Once
properly compiled and loaded you should see lspci list which kern
There are multiple ways to do time-of-day routing.
ExecIf w/ IFTIME, GotoIfTime, and ExecIfTime.
I put some examples below.
Sincerely,
Brian LaVallee
On 9/12/14, 10:05, Eric Wieling wrote:
See ExecIf in the output of "core show applications". The IF function might be useful,
see
This would NOT load)
; -- The parser stopped loading anything past the above mistake --
; -- Missing that space started a block-comment - Arghhh! --
exten => _4X.,1,NoOp(This would NOT load either)
; -end
Guess I have to change my highlight syntax, avoiding dashes in the future.
Sin
On 8/11/14, 11:31, Matthew Jordan wrote:
On Sun, Aug 10, 2014 at 5:02 PM, Deepak Rawat
wrote:
On Mon, Aug 11, 2014 at 3:29 AM, Deepak Rawat
wrote:
Hi,
I modified the query in res/res_config_odbc.c.
Original: "SELECT MAX(LENGTH(var_val)) FROM %s WHERE filename='%s'"
Modified: "SELECT MAX(
basic features
(hold, transfer, redial) are available by default. To duplicate the
digital PBX features you're looking for, will involve two groups of
settings. Configuration on the server -and- configuration on the phone.
SIP phones are NOT dumb terminals, you ha
ITE,ACK,CANCEL,BYE,REGISTER,PRACK,UPDATE
Accept: application/sdp
Sincerely,
Brian LaVallee
On 6/25/14, 11:30 PM, Rafael Visser wrote:
> Hi gurus!!!
>
> I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn
> Every minute asterisk sends an OPTION Request, i beleived
looking to manipulate the ISDN message via SIP, it all
comes down to how the gateway handles the desired functions.
Sincerely,
Brian LaVallee
On 6/26/14, 11:24 PM, Positively Optimistic wrote:
> We're using a Earthlink PRI converted to SIP via a MediaGateway. I assume
> the mediagateway wil
Hi Jonas,
While I don't work with queues, but you could playback announce-holdtime
before putting the caller into the queue.
exten => _X.,1,NoOp(Post Queue Announcement)
same => n,Answer()
same => n,Wait(10)
same => n,Playback(announce-holdtime)
same => n,Queue(real_queue
Since there are a number of setting that could be causing the alarm,
AMI/B8ZS, SF/ESF, etc...
Start with a loop-test, make sure the card can communicate with itself
(using the current settings).
Connect the following pins:
01 (RX-) <--> 04 (TX+)
02 (RX+) <--> 05 (TX-)
Sinc
7;ve though about passing the variable between the middle servers in a
SIP message, side communication channel. But, hoping there might be a
simpler solution.
Sincerely,
Brian LaVallee
--
_
-- Bandwidth and Colocation Provided b
it's connecting a sufficient number of PSTN connections to support those
users.
Sincerely,
Brian LaVallee
On 12/18/13, 11:45 PM, bilal ghayyad wrote:
Hello;
Can someone advise me what is the maximum number of users (IP Phones)
that can be supported by asterisk 1.8 or later?
Reg
Are you looking for something like this?
Note: This will continuously go between the two trunks until the caller
hangs up, can be fixed by adding loop counter.
;
; extensions.conf
;
[LOADBALANCE]
exten => _X.,1,NoOp(Connect to least used trunk)
; - show active count
exten => _X.,n,NoOp(Calls:
On Tue, 10 Dec 2013 23:02:45 +0200
Tzafrir Cohen wrote:
> On Wed, Dec 04, 2013 at 02:12:41PM -0500, Ruddy Gbaguidi wrote:
> > I never tought this is become a Linux vs Windows fight.
> > We have been using asterisk on linux from a long time now and happy
> > with it.
> > But some of our customers
it works for extensions and does NOT work on the
context field of the sippeers table, is there any field that can be used?
Sincerely,
Brian LaVallee
---===
;# extconfig.conf
;
[settings]
;
sippeers => mysql,database,sippeers
moresippeers => mysql,database,moresippeers
extensions => mys
e
way. In any case good luck with your project. If anyone else has more
recent experience regarding RTC please feel free to correct me. I'm
inclined to fiddle around with a VM based Asterisk install again if
it's gotten simpler to implement.
Brian
> On 22/11/2013 1:18 PM, Todd R. wro
the applications of this tool ? For example:
>
> run VoIP calls from scripts
> from web cgi pages
> from javascript in a browser window...
>
HTH and good luck.
Brian
[1]
http://www.linphone.o
lly a small unit that handles one or two DS3's.
The advantage comes when you add the 29th DS1. With VT1.5 it's just adding
a single channel, DS3 will require another whole DS3 to get an additional
DS1.
Sincerely,
Brian LaVallee
> From: Nick Khamis
> Reply-To: Asterisk Use
sterisk-users] Initial REGISTER Request: Contains Credentials
> before 401
>
> Brian LaVallee wrote:
>>
>> My SIP provider is not happy that credentials (in the Authorization header
>> field) are provided in the initial REGISTER request.
>>
>> The SIP provide
in the Asterisk community know how to avoid sending the
credentials until AFTER receiving a 401?
Any suggestions would be appreciated!
Sincerely,
Brian LaVallee
# ===
# sip.conf
# Asterisk 1.8.15-cert1
# ---
;
[general]
;
; - trucated
;
regis
firewall in place.
To add the Asterisk server to phpMyAdmin, see the following:
http://goo.gl/J1Py5
Brian
From: Lobna Hegazy
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Tue, 14 May 2013 22:57:34 +0200
To: Asterisk Users Mailing List - Non-Commercial Discussion
Thanks Jeremy!
>
> On 5/9/13 8:21 PM, Brian LaVallee wrote:
>> When qualify is enabled on a trunk, the From line shows "asterisk". See the
>> SIP message below.
>
> I had the same annoyance/issue. fixed it in
> https://issues.asterisk.org/jira/browse/
ld like to keep qualify enabled without sending the other end any
reference to "asterisk".
Can anyone point me to a setting that will change or remove `²asterisk²`
from `FROM:` in the OPTIONS message?
Thanks,
Brian LaVallee
--
/etc/asterisk/sip.conf (Asterisk 1.8.15-cert1)
[general]
yes, but thats no good, as the codec order will be ignored. I need to be able
to
allow Asterisk to choose the code from the order, without forcing a single
codec,
--
_
-- Bandwidth and Colocation Provided by http://www.api-dig
Hello
I use asterisk realtime, and I can set the order of codec preference on my
realtime allow column.
If I could disable transcoding, then I can always ensure a passthrough of the
common codec from origin to destination without transcoding (expensive on CPU)
-
and more or less, force the cod
Hi,
Is there any way to tell via the AMI or console if a given SIP channel is
hold? ChanIsAvail in the dialplan appears to have a 'hold' status, but
AMI and CLI commands tend to return 'in use', which is the same state as a
regular active c
ly specify their plan like this:
"1[2-9]x|[2-9]x|60[2-9]|6[1-2][0-9]|63[0-2]|70[0-9]|71[0-5]"
Is there a benefit to specifying the full dial plan? Or should I just
use the smaller plan that matches everything?
Thanks,
Brian
--
On Sun, Feb 12, 2012 at 12:59 AM, Bruce B wrote:
> If your server is open to the internet and in SIP general section you have
> nat=no and in peers you have nat=yes or vice versa then it's possible to
> enumerate your extension. Because Asterisk responds with different messages
> if the extension
so, did you have to make
> any changes to the SIP header sent to make Polycom phones auto answer? ***
> *
>
> ** **
>
> Regards,
>
> ** **
>
> Mike
>
> **
>
Hi
;
Hi Danny,
http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom
Brian
> --
> www.danntel.net
> *sip:danny4...@thesipschool.com*
> sip:dann...@opensips.org
>
>
>
>
>
> --
> _
>
I am trying to use the FILTER() function to strip out "/" from a CID
name. I have the following in my extensions.conf where I want to
perform the filtering:
exten => s,n,Set(NAME=${FILTER(\x20-\x2e\x30-\7d,${DIAL_NAME})})
However, when ${DIAL_NAME} is, say, "J & J DOE" the string resulting
from
and the repair work was of questionable
quality. Also our service entry point / punch-down area is a rat's nest
(one building and service is shared by three companies). I guess I can
chalk this behavior up to the wiring.
ead. Perhaps if the
caller happens to hang up right between the two Dial() commands?..
As an aside, the Set(CALLERID...) bit doesn't work. The idea was to prepend
a 9 so that a SIP user could use the "redial" feature of the phone's call
log to return a missed call (automatically
em: 2.0.95:beta April 2010
asterisk: Asterisk 1.6.2.9-2+squeeze1
OS: Debian Squeeze 64 bit
~# uname -a
Linux ps-pbx 2.6.32-5-amd64 #1 SMP Wed Jan 12 03:40:32 UTC 2011 x86_64
GNU/Linux
These are all unmodified packages obtained via aptitude.
What am I getting wrong?
Many thanks,
~Brian
---
rmation be available to a custom
application (AGI, etc)?
I am trying to create one-button access (via speed dial, e.g.) to more
complex Asterisk functions such as call park for our less technically savvy
reception employees.
Many than
t be happy without lights.
Cheers,
~Brian
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell
Sent: Tuesday, March 08, 2011 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
needed by FXS
modules).
I've tried google searches but haven't found anything mentioning this odd
behavior. Is this expected?
Many thanks,
~Brian Henning
------
Brian Henning, Software Engineer
/\Pine Research Instru
even after it's lost registration.
It *looks* like the problem is the innomedia since it didn't send
another register. But I figured I'd ask to see if anyone here knew what
the problem could be. Otherwise
switch it back and reboot again.
Does anyone know how to setup this phone to work with asterisk so that
the indicator light comes on when there's a new message and goes off
quickly (less than a minute) after the message is deleted?
Thanks,
Brian
--
The subject says it all. I'm betting there's a way to do it, but so far
I haven't found the dialplan runestone via web searching.
Thanks.
b.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to As
I wonder if anyone out there has a perspective on this. There are a
welter of tickets out there on the matter, most of them closed.
This problem began for me over a year ago, and continues up to the
latest versions I've installed (1.6.2.13).
It happens randomly, and the suggestion on one of th
I'm still working on it, but I am using a2billing and making
modifications to some of the PHP code. I modified the layouts of their
default invoices and and added PDF creation using dompdf
(http://code.google.com/p/dompdf/downloads/list).
-b
On 08/03/2010 09:41 AM, Zeeshan Zakaria wrote:
I currently have 4 lines coming into the house. We currently have an Avaya
standard analog key system which has served us well, but running extensions is
a major pain and requires a dedicated run per extension. I have ethernet run
throughout the house though.
The first two lines are "home" l
Create a local mail alias that sends to what you need and then use the
alias in the vmail config.
-b
On Tue, 2010-06-01 at 20:47 +0200, Jonas Kellens wrote:
> I am no programmer, and very happy with what Asterisk holds in it.
> Just hoped that mailing multiple mail-addresses was an easy
> configu
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