In the forthcoming 1.4, you can tell the Queue application to run an AGI just
before sending the call to the destination. In the AGI, you can use the (also
new in 1.4) MEMBERINTERFACE channel variable to determine the destination.
Of course, that's not a solution now since 1.4 is not even
This is actually working as designed. You need to use type=peer in order for
call-limit to work properly, which in turn is what allows hints to work
properly.
- Brad
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Gagnon
Sent: Tuesday, August
If you already have the IP in a file, why don't you set it up so the
file itself says: externip=xx.xx.xx.xx and then do a #include in
sip.conf for the /etc/myip file? I believe you'll have to do a sip
reload either way (which can obviously be part of your cron job) if
you're not already, but
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Wednesday, August 23, 2006 8:40 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hint extension issue - bug?
On Wednesday 23 August 2006 06:56, Watkins, Bradley wrote:
This is actually
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Wednesday, August 23, 2006 9:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hint extension issue - bug?
On Wednesday 23 August 2006 08:48, Watkins, Bradley wrote:
It's not a bug. When you use type
I may have to eat my words, then. This is the case with trunk, and I can't
recall the last time I built a 1.2.x system. I could have sworn that behavior
didn't change, but I've been wrong before.
- Brad
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf
Only the Asterisk box that a phone is registered on WILL send the sip
notify messages. The others will have no idea where to send them, and
will not do so.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Tuesday, August
Message-From: Watkins, Bradley
[mailto:[EMAIL PROTECTED]Sent: Thursday, August 03,
2006 2:06 PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: RE: [asterisk-users] Re: DUNDi with
SIP
I forget if this does what you want, but try adding a fromuser
setting to yoyr
: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: Chocolate Chip sip:[EMAIL PROTECTED];privacy=off;screen=no
Content-Length: 0
--- (11 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
-Original Message-
From: Watkins, Bradley
[mailto:[EMAIL PROTECTED] Behalf Of Watkins,
Bradley
Title: RE: [asterisk-users] Re: DUNDi with SIP
I forget if this does what you want, but try adding a fromuser setting to yoyr peer entries.
- Brad
-Original Message-
From: Douglas Garstang [mailto:[EMAIL PROTECTED]]
Sent: Thu Aug 03 15:58:50 2006
To: Asterisk Users Mailing List -
The way to make this work is to define a sip user/peer with the IP
address in it, then have your dundi.conf entry look like:
180netsip = global_dundi_local,1,SIP/peername/${NUMBER},nopartial
As far as I can tell from the code, this is the only way to make it work
properly based on the way the
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday, August 02, 2006 6:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
-Original Message-
From: Watkins
Try putting a username= in the peer (BTW, use peer not friend)
definitions. You appear to be attempting to authenticate as the
originating callerid (3254101).
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday,
: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Wednesday, August 02, 2006 5:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
Try putting a username= in the peer (BTW, use peer not friend)
definitions. You appear
When we were first looking at Asterisk, I explored some options of
integrating our existing Octel voicemail systems with it. The only
possible way I could come up with (understanding that I am by no means
an Octel expert) was DTMF inband integration. The most difficult part
seemed to be
No. The GPL does not require installation of source by default, only
that it be made available upon request. If this were the case, then
*every* GNU/Linux distro I know of would be in violation at least for
the kernel and certainly for other things (all the GNU tools).
Regards,
- Brad
I can say with absolute certainty that in our installations
using the blind transfer of the Polycom (NOT the Asterisk transfers) will show
the original caller ID and not the caller ID of the transferer. Attended
transfers, of course, show the transferer since that is a new call
initially.
. The Polycom
doesn't give you the second transfer button, to release the call as unattended
until AFTER the destination number has started to ring.
Doug.
-Original Message-From: Watkins, Bradley
[mailto:[EMAIL PROTECTED]Sent: Tuesday, July 25, 2006
4:15 PMTo: Asterisk Users Mailing List
I recently got this going, and had a similar experience. In my case,
the solution was to set the RPID to the expected number assigned to the
account. YMMV, but it's worth a try.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kharris
Yes, use the bonding driver. That way you only have one IP address and
both connections are viewed as one logical.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of shadowym
Sent: Thursday, July 20, 2006 12:45 PM
To:
This is the tact that I take, and it's never been a problem for us.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Monday, July 17, 2006 2:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Could you possibly put up the relevant section(s) of your sip.conf? It sounds
like the DUNDi portion is set up properly, and obviously it's not going to find
an extension that doesn't exist.
Regards,
- Brad
From: [EMAIL PROTECTED] on behalf of Simon Woodhead
My first advice is to double and triple-check that your
switchtype settings match. We had an issue that looked *exactly* the same
from a PRI debug standpoint and it turned out that while the carrier said we
were NI2, it was actually DMS100.
Regards,
- Brad
From: [EMAIL PROTECTED]
I actually have a server that *may* have exhibited the same
problem. I can't say for certain as the system itself was acting a bit
wonky, but I can say for certain that after we disconnected the PRI (actually,
switched it over to the secondary box in our cluster) that users connected to
I have definitely run into this on the one production site I have with
1.2.9.1
I haven't tried backrevving in order to see if it affects 1.2.9 or
older, but it is very annoying.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
I'd certainly be up for it, even if it ended up being a small group that
met over beer at a local pub. :)
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of BerkHolz,
Steven
Sent: Thursday, June 22, 2006 4:27 PM
To:
I would say it's almost certainly a cabling issue of some sort. We get
the S-frame while link down message all the time on our Asterisk
clusters due to the way the T1 failover switch works. It's harmless in
our case since really the PRI is connected to the other box in the
cluster (unless there
Unless I'm misunderstanding what you're looking to do, Aaron has hit the nail
on the head here. You need to set it up so that the secondary, tertiary, etc.
boxes are weighted differently. That way, you need not know or care about the
weights directly within the dialplan.
Regards,
- Brad
Polycom 601, hands down.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Warren
Sent: Thursday, June 15, 2006 10:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Best $300 VoIP phone for asterisk?
If
the 501? I do not have PoE here
(at least not yet) and do not need the additional lines (except maybe on
the attendant's console).
Is the 601 better than the 501 or does it just have the ability to
handle more lines?
W
Watkins, Bradley wrote:
Polycom 601, hands down.
- Brad
-Original Message
-
From: Watkins, Bradley
[mailto:[EMAIL PROTECTED] Behalf Of Watkins,
Bradley
Sent: Thursday, June 15, 2006 2:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Not Able to Handle Complex
FailoverSituations
Unless I'm misunderstanding what
: Thursday, June 15, 2006 2:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Not Able to
HandleComplexFailoverSituations
-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 10:36 AM
To: Asterisk
Yes, what is it you attempting? I use DUNDi extensively, though you are
correct that the existing docs don't go very far in describing some
things.
Regards
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Wednesday, June 14,
Title: Message
The
first situation you mention can be solved by creating separate files that
contain the unique elements, and then including them in the main files where all
the commonality is. That is how we do things, and it works well for
us. It may be a little cumbersome if you have a
I've never attempted to use this feature, so I can neither confirm nor
deny whether it works/doesn't work/used to work/etc.
But what I find really odd, is that the code doesn't even appear to try
and parse astdb when it's loading the config, at least insofar as I
can tell. A quick grep -i astdb
making it into chan_sip.c
However, the option *did* make it's way into sip.conf, so I guess that
the real bug is that the option is in sip.conf.
Bummer.
Devels: Any chance of getting 3359 re-opened and put into asterisk ?
Julian
Watkins, Bradley wrote:
I've never attempted to use this feature, so
Yes, the (newly rewritten, but compatible with older AEL files) version
of AEL that is currently in trunk is staying and I think personally it's
a big step forward. I still haven't gotten used to it myself, but then
the folks who used to write everything in assembler probably took a
while to get
I have not found a way to do this via the Polycom
configs. However, what I do is just ensure that the callerid of an inbound
call is set so that the recorded number on the Polycoms is a valid callback
number (i.e., prepend '9' or '91' depending on the inbound
CallerID).
Regards,
- Brad
How about the dialplan function SIP_HEADER?
-= Info about function 'SIP_HEADER' =-
[Syntax]
SIP_HEADER(name)
[Synopsis]
Gets or sets the specified SIP header
[Description]
Not available
dtw-test-asterisk-001*CLI
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
I'm not sure if you have considered this, but if you were using SIP
between the Asterisk servers you can definitely achieve this using
X-headers.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, May 11,
The answer is to create a separate context for your regextens (or, more
appropriately, name it in sip.conf and let chan_sip create it) and then
include that context in your dundi_local context where you have the
dialing information.
Regards,
- Brad
-Original Message-
From: [EMAIL
Not to mention the obvious, and this may not help your situation, but if
you were (or are) using templates it would be a one-line change.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of William
Piper
Sent: Tuesday, May 02, 2006 12:49 PM
The OP was referring to how sox interprets filename extensions. In that
case, Kevin's .raw and .sw extensions are correct.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, April 19, 2006 12:56 PM
To:
That implies that the 2850 has a standard molex connector anywhere inside of
it, which is not the case.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Davies
Sent: Wednesday, March 29, 2006 6:34 AM
To: [EMAIL PROTECTED]; Asterisk
but
when I call the users above 70 it prompts me User Not Found.
Any idea regarding this?
Thanks,
Ryan
At 04:12 AM 3/24/2006, Watkins, Bradley wrote:
I don't think there's any kind of (significantly small, anyway) limit.
I have over 300 users at one site in voicemail.conf and no issues
I don't think there's any kind of (significantly small, anyway) limit. I
have over 300 users at one site in voicemail.conf and no issues there.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ryan Pagquil
Sent: Wednesday, March 22, 2006
I apologize, but the fix I was thinking of wasn't directly related to this.
It was in app_voicemail.c, but related to using the channel's context for
the Directory application. The fix for your issue may be indirectly
related, though. I would open a bug.
Regards,
- Brad
-Original
Did you download it from asterisk.org? I didn't have the latest -addons,
but I just downloaded it and it does have Copying which contains the GPLv2.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. J.
Meidlinger
Sent: Wednesday, March
What version of Asterisk are you running?
The reason I ask is that I think I remember a fix for this on the
svn-commits list awhile back.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson
Sent: Tuesday, March 21, 2006 4:52 PM
No flames here as I realize that there are plenty of limitations with MySQL,
but if you're using the current GA of it views is not one of them.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton
Sent: Sunday, March 19, 2006 4:15
At the moment I'm out of the office, but when I return I'll be certain to do
that. Note that my solution is different from what you are working on with
regexten, though I suspect some of the challenges that I've faced and
overcome are not. I'm actually using UltraMonkey for load-balancing and
problem when
loadbalancing with Ultramonky? As I understand it LVS does not properly
support SIP in that it doesn't always use the same source path.
regards,
David
On 3/17/06, Watkins, Bradley [EMAIL PROTECTED] wrote:
At the moment I'm out of the office, but when I return I'll be certain
to do
.
Regards,
- Brad
_
From: [EMAIL PROTECTED] on behalf of David Thomas
Sent: Fri 3/17/2006 3:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: DUNDi Halfway and CLUSTERING
On 3/17/06, Watkins, Bradley [EMAIL PROTECTED] wrote:
Do
).
Regards,
- Brad
_
From: [EMAIL PROTECTED] on behalf of David Thomas
Sent: Fri 3/17/2006 5:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: DUNDi Halfway and CLUSTERING
On 3/17/06, Watkins, Bradley [EMAIL PROTECTED] wrote:
I
Could you perhaps post your dundi.conf for both boxes? I'm afraid this
message doesn't mean anything to me, but I have about a dozen boxes doing
DUNDi peering so I know what the config should look like. But it's
basically always worked for me.
Regards,
- Brad
-Original Message-
From:
I have several servers using them, but I only needed to download them
directly from you just once. I replicated the bits myself. The magic of
these advanced technologies... ;)
I could go download them a few more times if it would make you feel better.
All kidding aside, I don't think I ever
That depends on what you mean by default. The supplied sample
extensions.conf contains the priorityjumping=no by default, but if this
parameter is absent then the default is to jump n+101.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
It must be microseconds that is being quoted, as even the 2626 that you
mention lists a less than 13.3 microsecond latency.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Ankers
Sent: Thursday, February 23, 2006 6:54 PM
To: 'Asterisk Users
While I've never actually tried exactly what you're doing below
(constructing a variable name from strings and other variables), it looks
like the variable substitution you're attempting is not being done properly.
Try something like:
exten = s,3,GotoIf($[ ${NUM${mainLoop}_CMD} = Dial ]?5:7)
Title: Message
It
looks like the outbound caller ID is not being set properly. Most of the
carriers that I've dealt with will act exactly as you said if you do not set it
to what is expected at the 911 center.
In
particular:
Calling Number (len= 8) [ Ext: 0
TON: Subscriber Number (4)
For your second question, how about the application MailboxExists? You
could write a quick front-end that asked for the mailbox and then used
MailboxExists to test. If it doesn't, perhaps increment a counter (so you
can disconnect later if it exceeds some value) and then return to the
original
I had this same issue with 601s, and I was able to fix it by defining:
progressinband=yes in sip.conf.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio
Sent: Tuesday, January 31, 2006 11:20 AM
To:
I can say that I've implemented it on several Asterisk servers using the
802.3ad mode and it works very well. The failover is quick, and there are
none of the issues mentioned here. I'm not particularly concerned about
running at GigE speeds as the level of traffic in/out of these boxes is
The filename needs to be MAC Address-directory.xml, not IP
Address-directory.xml. Grab the MAC off the back of the phone.
This is the same as for the provisioning files if you are using your TFTP
server to do that. Also, is there a reason that you aren't using FTP? It's
much more robust, and
, Watkins, Bradley [EMAIL PROTECTED] wrote:
The filename needs to be MAC Address-directory.xml, not IP
Address-directory.xml. Grab the MAC off the back of the phone.
This is the same as for the provisioning files if you are using your
TFTP server to do that. Also, is there a reason that you aren't
I believe the odbcstorage variable in voicemail.conf needs to be set to the
name of the connection in res_odbc.conf not the dsn. So in your example,
you would need: odbcstorage=MySQL
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
The only time I've ever received that message was when I was not receiving
CallerID on the line (though this was with a Rhino channel bank).
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Tuesday, November 15,
I would certainly be interested in this and could perhaps help out. I'm
currently building a heartbeat/LVS/ldirectord load-balanced cluster that
could use some significantly better health checking(which was my next step).
Since ldirectord is already written in perl, it would avoid having to call
What Steve posted is the output of the command 'show application RxFAX' on
the Asterisk CLI.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Woody Sturges
Sent: Wednesday, August 31, 2005 11:15 AM
To: Asterisk Users Mailing List - Non-Commercial
Title: Message
I
believe you have to upgrade to 5.3 in order to go from unsigned to signed
executables. Once you're at 5.3, you can go directly to 7.5. I did
this recently with a couple of 7960s I had in the lab and it worked
perfectly.
Regards,
-
Brad
-Original
What revision of card is the new one? It sounds like you have one of the
new Rev I cards and you aren't running either 1.0.9 or CVS HEAD. Either of
these will solve your problem if I am correct.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
The only phone that I know of that has a 10/100/1000 switch in it is the
Cisco 7971G-GE:
http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0900a
ecd801c5c4a.html
It is one heck of a phone, but the price is relatively astronomical (list
price is roughly $1100 each).
- Brad
Ooops... I should also mention that apparently they don't support SIP (I
was just looking). I saw them demoed by Cisco on a CallManager box awhile
ago. I guess I just assumed that a newer phone like that would have SIP
firmware available as well.
Sorry for any confusion.
- Brad
-Original
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