That was quite an interesting set of responses. I didn't get any
impression that there is a strong preference either way.
Thanks for all the replies.
Al
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asterisk-users
I have added t in dialplan
exten = 1234,1,Dial(USTM/2...@c,40,t)
so now i can transfer, but when the caller the extension I transfer to hangs up
asterisk dumps an I have to start it up again.
Any thoughts?
/ralf
From: asterisk-users-boun...@lists.digium.com
9 feb 2009 kl. 23.17 skrev Raj Jain:
On Mon, Feb 9, 2009 at 4:43 PM, Olivier oza-4...@myamail.com wrote:
Hi,
My patton 4638 is sending :
v=0
o=MxSIP 0 46 IN IP4 192.168.100.52
s=SIP Call
c=IN IP4 192.168.100.52
t=0 0
m=audio 4984 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101
On Wed, Feb 11, 2009 at 2:49 AM, Steven J. Douglas stev...@moij.biz wrote:
Hi,
Have you tried using externip in your sip.conf? By setting the correct
localnet, any SIP packets that goes elsewhere will use the value in
externip. This might solve your problem.
Regards,
Steve
yes i've done
Hi all,
I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is
OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy
is not replying back...The issue is the UNKNOWN username that reside in the
OPTIONS packet as you can see in the captured packets as you
Try manipulate the fromuser= parameter in sip.conf.
On Wed, 11 Feb 2009 11:43:13 +0200, michel freiha mich...@gmail.com
wrote:
Hi all,
I need to register asterisk on an OpenSIPS SIP Proxy...The Registration
is
OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP
Proxy
is
I would like to know if I can set Call Forwarding on an extension but allow
direct calls from the extension it is forwarded to.
Example:
Extension 100 sets call forwarding (all) to extension 101.
All calls to 100 are immediately forwarded to 101 as expected.
However, if 101 tries to transfer
Vieri schrieb:
I would like to know if I can set Call Forwarding on an extension but allow
direct calls from the extension it is forwarded to.
Example:
Extension 100 sets call forwarding (all) to extension 101.
All calls to 100 are immediately forwarded to 101 as expected.
However,
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201 |
awk '{ print $1 '} )
It basically calls *, gets the list of channels, filters them out to get the
channel name and
--- On Wed, 2/11/09, Philipp Kempgen philipp.kemp...@amooma.de wrote:
I would like to know if I can set Call Forwarding on
an extension but allow direct calls from the extension it is
forwarded to.
Example:
Extension 100 sets call forwarding (all) to extension
101.
All calls
Philipp Kempgen schrieb:
Vieri schrieb:
I would like to know if I can set Call Forwarding on an extension but allow
direct calls from the extension it is forwarded to.
Example:
Extension 100 sets call forwarding (all) to extension 101.
All calls to 100 are immediately forwarded to 101 as
On Wed, 11 Feb 2009, Alan Lord (News) wrote:
That was quite an interesting set of responses. I didn't get any
impression that there is a strong preference either way.
I asked the same question some time back too... Got a few replies, and now
(as then), all my systems are 100% .conf (or
--- On Wed, 2/11/09, Philipp Kempgen philipp.kemp...@amooma.de wrote:
I would like to know if I can set Call Forwarding
on an extension but allow direct calls from the extension it
is forwarded to.
Example:
Extension 100 sets call forwarding (all) to
extension 101.
All calls to 100
Gordon Henderson schrieb:
On Wed, 11 Feb 2009, Alan Lord (News) wrote:
That was quite an interesting set of responses. I didn't get any
impression that there is a strong preference either way.
I asked the same question some time back too... Got a few replies, and now
(as then), all my
--- On Wed, 2/11/09, Vieri rentor...@yahoo.com wrote:
I would like to know if I can set Call
Forwarding
on an extension but allow direct calls from the
extension it
is forwarded to.
Example:
Extension 100 sets call forwarding (all) to
extension 101.
All calls to 100 are
I'm a big fan of Audicodes MP-124. Very stable and a full feature set,
and it has amphenol connectors so you can tie directly to 66-blocks for
cabling your phones.
j
On Tue, 10 Feb 2009, Erick Perez wrote:
Hi, I am looking to connect 66 analog phones to an asterisk box. I was
thinking of
svn co https://agx-ast-addons.svn.sourceforge.net/svnroot/agx-ast-addons agx-ast-addons
./build_sh from the trunk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: 10 February 2009 18:35
To:
On Wed, 11 Feb 2009, Philipp Kempgen wrote:
Gordon Henderson schrieb:
On Wed, 11 Feb 2009, Alan Lord (News) wrote:
That was quite an interesting set of responses. I didn't get any
impression that there is a strong preference either way.
I asked the same question some time back too... Got a
Hello,
We are looking for someone who can act as our 'remote Asterisk hands' in
Mexico. One of our customers has opened an new office in the 'Col. San
Rafael, Delg. Azcapotzalco' region. We need someone (or organisation) to
perform the onsite installation and connections for an Asterisk server
Vieri schrieb:
I'm using Set(RETURN_EXT=${BLINDTRANSFER:4:4})
but that assumes that I have only 4-digit extensions
Well, skip the length argument (the second :4).
and all have the same prefix (SIP/).
Is there a more portable way?
Set(RETURN_EXT=CUT(BLINDTRANSFER,/,2));
Philipp Kempgen
Hi helpers,
I seem to have a problem of intermittent DTMF tones being played during
a conversation.
Eg: Extn 100 takes an inbound call and all is fine. Except, at an
undetermined time the person on extn 100 will here a DTMF tone for no
apparent reason (it's not the caller pressing buttons).
David @ULC schrieb:
Looking for a Free VOIP Billing and Soft Switch.
soft switch includes back-to-back user agents (Asterisk) I guess?
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk: http://the-asterisk-book.com -
Looking for a Free VOIP Billing and Soft Switch.
Any suggestions ?
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Andrew Thomas wrote:
I seem to have a problem of intermittent DTMF tones being played during
a conversation.
I'm having the same problem, but in my case, it's every 1 minute and at
the start of the call.
I wonder if it has anything to do with echo cancellation.
I've only noticed when using a
On Tue, Feb 10, 2009 at 12:23 PM, Erick Perez eaper...@gmail.com wrote:
Hi, I am looking to connect 66 analog phones to an asterisk box.
other hardware suggestions for this task will be nice.
Citel makes a box:
http://www.citel.com/Products/Portico.asp
If you're converting from Definity,
Don't expect too much from Aastra. In our previous dealings trying to
report serious bugs (like phone lockup/crash) to Aastra, they didn't want
the details, or they simply gave us canned answers which did no good.
(Superficial tech support)
We've moved away from Aastra for new installs, but we
2009/2/11 OCG Technical Support supp...@ocg.ca:
Don't expect too much from Aastra. In our previous dealings trying to
report serious bugs (like phone lockup/crash) to Aastra, they didn't want
the details, or they simply gave us canned answers which did no good.
(Superficial tech support)
On Wednesday 11 February 2009 08:22:39 Gordon Henderson wrote:
For my appication, I get on OK with pure dialplan. I have a fully
featured PBX system which runs on nothing more than dialplan, and I'm
happy with it. I do have something higher level that generates some of
the dialplan for me, but
On 11 Feb 2009, at 14:22, David @ULC wrote:
Looking for a Free VOIP Billing and Soft Switch.
And you are asking an Asterisk list... Asterisk? Billing is probably
best doing a custom job..
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Hi men,
Resolved for one of my customers by upgrading Asterisk/Libpri/Zaptel.
I don't remember what wer the versions, sorry.
Check and advise us the results, please.
Best Regards,
Francois
No virus found in this outgoing message.
Checked by AVG - www.avg.com
Version: 8.0.233 / Virus Database:
David @ULC wrote:
Looking for a Free VOIP Billing and Soft Switch.
Any suggestions ?
I'm looking to put the milk back in the cow.
If you have the skinny on that, maybe we can swap suggestions.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678)
Of course you should be using .conf
On Tue, Feb 10, 2009 at 2:28 AM, Lee, John (Sydney)
john@compuware.com wrote:
Of course you should be using AEL.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of
I have an extension defined like this:
exten = do_monitor,1,Answer()
exten = do_monitor,n,NoOp(Just got '${CfMC_ActionID}')
exten = do_monitor,n,ChanSpy(${CfMC_WhoHear},qX)
exten = do_monitor,n,Hangup()
I use an AMI packet like this:
Action: Originate
Channel: Agent/1001
Exten: do_monitor
Of course you should be using Lua.
More seriously, use whatever works best for you. If you have time,
evaluate all three alternatives and pick the one you like the best. If
you don't have the time, I wouldn't put a lot of effort into switching
to AEL or Lua based dialplans.
There are
Excuse my ignorance but if i have an asterisk in a LAN, and i have
users in their homes/internet (dozens), in order to correctly connect
those users across my firewall, what is the technology that i need to
buy, called?
secure border gateway?
session controller?
secure gateway?
the audiocodes site
Hi,
I'm trying to integrate the following into my Asterisk environment.
BPT Targa Single button audio panel
(http://www.bpt.co.uk/entry-control/pictures/albums/targha/pages/HSC1ST%20Targha%20Entry%20Panel%20Mounted_JPG.htm),
linked to an IT200 interface unit
OpenVPN?
--Tim
- Erick Perez eaper...@gmail.com wrote:
Excuse my ignorance but if i have an asterisk in a LAN, and i have
users in their homes/internet (dozens), in order to correctly connect
those users across my firewall, what is the technology that i need to
buy, called?
secure
It all depends on how much money you want to spend and how scalable you
want your platform to be, as well as your level of comfort with open source
technology stacks vs. proprietary vendor gear.
You could pull this off with a SIP proxy like Kamailio/OpenSIPS and
Mediaproxy if you wanted. And up
On Wed, 11 Feb 2009, Erick Perez wrote:
Excuse my ignorance but if i have an asterisk in a LAN, and i have
users in their homes/internet (dozens), in order to correctly connect
those users across my firewall, what is the technology that i need to
buy, called?
secure border gateway?
session
Of course you should be using Lua.
I really have to try that sometime
tAt the same time, traditional .conf dialplans
are not going away anytime soon, and you do not lose any functionality
vs AEL and Lua.
My reason for sticking with .conf files so far? dialplan show -- it
is easier
On Wed, 11 Feb 2009, Tilghman Lesher wrote:
On Wednesday 11 February 2009 08:22:39 Gordon Henderson wrote:
For my appication, I get on OK with pure dialplan. I have a fully
featured PBX system which runs on nothing more than dialplan, and I'm
happy with it. I do have something higher level
Hi All;
Why the below does not work? Since about 10 days?
wget http://ftp.digium.com/pub/zaptel/zaptel-1.2-current.tar.gz
Regards
Bilal
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To
Matthew Nicholson schrieb:
Of course you should be using Lua.
I'd like to file a bug report.
Incited by all this extensions.* euphoria I'm trying to use
extensions.js but Asterisk doesn't even try to read the file.
The logs don't give any helpful information about the problem.
Reproducibility:
On Wednesday 11 February 2009 13:08:54 bilal ghayyad wrote:
Hi All;
Why the below does not work? Since about 10 days?
wget http://ftp.digium.com/pub/zaptel/zaptel-1.2-current.tar.gz
The server isn't down. The name of the server has been downloads.digium.com
for quite some time, and the 1.2
On Wednesday 11 February 2009 13:07:16 Gordon Henderson wrote:
On Wed, 11 Feb 2009, Tilghman Lesher wrote:
On Wednesday 11 February 2009 08:22:39 Gordon Henderson wrote:
For my appication, I get on OK with pure dialplan. I have a fully
featured PBX system which runs on nothing more than
Tilghman Lesher schrieb:
On Wednesday 11 February 2009 13:07:16 Gordon Henderson wrote:
On Wed, 11 Feb 2009, Tilghman Lesher wrote:
My viewpoint is that you should work on separation of your application
code versus data, so that other than new development, your dialplan
should be
On Wednesday 11 February 2009 13:39:58 Philipp Kempgen wrote:
Tilghman Lesher schrieb:
On Wednesday 11 February 2009 13:07:16 Gordon Henderson wrote:
On Wed, 11 Feb 2009, Tilghman Lesher wrote:
My viewpoint is that you should work on separation of your application
code versus data, so
On Wed, Feb 11, 2009 at 12:14:33PM -0600, Matthew Nicholson wrote:
Of course you should be using Lua.
More seriously, use whatever works best for you. If you have time,
evaluate all three alternatives and pick the one you like the best. If
you don't have the time, I wouldn't put a lot of
Howdy,
Working on some niche requests from one of my hotel clients. asterisk
1.4.20-1 on CentOS, Polycom 501s.
The first request is for the Polycom's screen to show the CID of the
inbound caller when a call pick is executed, so the picker knows if the
call is internal or external. I have
Jeff LaCoursiere schrieb:
Working on some niche requests from one of my hotel clients. asterisk
1.4.20-1 on CentOS, Polycom 501s.
The first request is for the Polycom's screen to show the CID of the
inbound caller when a call pick is executed, so the picker knows if the
call is internal
When action_userevent was rewritten to not use local variables there was an
omission. The buffer is not initialized each time so things keep getting
appended to the buffer.
In addition I would find it useful to have the ping action return the
timestamp. That way I do not have to have timestamp
I use channel banks.
I like the Adit 600s. For your configuration you'd need 2 Adit 600
with 9 FXS cards and 3 T1 ports in the Asterisk box.
One side advantage is that you can mount the Adit 600 right next to
your cat3 wiring, then just use an existing cat3 to the Asterisk box.
I have seen lots of
Greetings list,
Wondering if anyone has come across this strange dialplan pattern matching
issue before:
I have a context defined as follows (the plus simply implies it follows on from
an existing context in another #include - which, yes, has been included first):
[privatedundi](+)
exten =
On extensions.conf.sample I see this:
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a '_'
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from
What is IDAP-T1? How different is it from normal T1?
Any chance I can get it to work with Digium 412P and Asterisk 1.4.* ?
If yes, what would zaptel.cof look like?
Any difference from normal T1 config?
Thanks.
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--
Nabeel Jafferali
X2 Networks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: February-11-09 8:18 PM
To:
BTW I just did some quick experimentation. Example 1 did not work, example 2
did work. So that's a solution to your issue.
Example 1:
exten = 999,1,Goto(nabeel,1)
exten = _nabeel,1,Goto(800,1)
Example 2:
exten = 999,1,Goto(nabeel,1)
exten = _[n]abeel,1,Goto(800,1)
--
Nabeel Jafferali
X2
Hi,
As others have mentioned, the 'n' is a pattern char.
I have a system that uses similar tricks to yours. What I did about
this issue was to change the pattern match chars to be upper case
only. Drop me a line if you want the patch.
Regards,
Steve
On 2/12/09, Chris Bagnall
Everybody is talking about other products. But yes, the Xorcom will
handle all ports active, supports a high density connector at the
back, looks just like standard Zap/Dahdi ports to Asterisk, rack
mounts nicely and much less $$ than the other solutions.
Steve
On 2/10/09, Erick Perez
What is Zap mirroring ?
- Original Message -
From: Danny Nicholas da...@debsinc.com
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Monday, February 09, 2009 10:09 PM
Subject: Re: [asterisk-users] How to make the Asterisk-GUI
Do you have extension ontext 059*162*178*122*78600051 in your
extensions.conf under the default context ?
- Original Message -
From: Philipp Kempgen philipp.kemp...@amooma.de
To: Asterisk Users asterisk-users@lists.digium.com
Sent: Monday, February 02, 2009 10:40 PM
Subject: Re:
I go with what I know is solid, Adit or Adtran channel banks and T1
ports. I personally like Adtran but that is just preference.
The Xorcom device is USB correct? I just have a personal block on
putting anything mission critical on a USB port or hub. I would have
to lab it up and really test
On Tue, Feb 10, 2009 at 01:56:16PM -0800, Mik Cheez wrote:
I use them both; my legacy dialplan is all .conf and new stuff is .ael.
I find AEL to be the better option when jumping around, but that's
just my opinion.
But isn't AEL just converted into .conf language anyway? Or has this
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