Hi,
I have recently run into the problem with macro implementation in AEL in
Asterisk 1.6. I have some older AEL dialplan which runs on 1.4 but it does not
on 1.6 and I'm not sure how to solve this correctly. Let me explain...
For example, in Asterisk 1.4 I have a macro like this:
Hi,
I'm not sure whether this is possible but if it is, I'm sure someone on
here might know ...
Is it possible to use Monitor() to record a conversation[1], but make it
start a new pair of wav files at intervals (eg every 15 minutes) if the
calls go on for a long time?
We already have this
I have had a couple thunderstorms take out a card, again last night.
The card with dahdi show status still report OK both times.
When calling into the card I get all circuits are busy.
However, simply replacing the card did the trick. Before that
I stopped asterisk, restarted DAHDI, rebooted
Jerry Geis wrote:
I have had a couple thunderstorms take out a card, again last night.
The card with dahdi show status still report OK both times.
When calling into the card I get all circuits are busy.
However, simply replacing the card did the trick. Before that
I stopped asterisk,
In article 4ec120c1.8080...@pagestation.com,
Jerry Geis ge...@pagestation.com wrote:
I have had a couple thunderstorms take out a card, again last night.
The card with dahdi show status still report OK both times.
When calling into the card I get all circuits are busy.
However, simply
Am 14.11.11 06:54, schrieb Linux:
I tried to understand the rfc4235 which states the following:
However, using this package to model state for non-
session dialog usages is out of the scope of this specification.
Does this actually mean that the device state of being offline is
Hello Stefan,
Thank you for your answer. I was already afraid it is not in the standard. I
will look into the custom device states.
thanks,
Hans
-Original message-
To:Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com;
From:Stefan Schmidt
Hi,
I am using the Dial-Application and need to hangup (caller and
callee) with more than one key. Is it possible to set this feature,
so that the caller or callee can hangup by pressing any of the keys
1, 2 or 3?
I tried to configure it in the context
Once the call is completed you can use SOX to split the call. In my
opinion, you will have to get a larger ram disk or record the files to a
different format like WAV49, but maybe somebody has a better solution for
you.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi.
Yeah, sox and soxmix are no problem - we're already using that to
merge/join all of the segments together if people pause then resume the
recordings mid call.
The main issue is getting Asterisk to split the recordings into segments
even when users don't pause/resume the recordings (which
I know this questions is not really asterisk related, however I know a lot of
people here are in the industry.
I was curious if anyone here could provide insight on how to become a
facilities based CLEC.
I did a lot of google-ing and read info on voip-info.org but it's all the same
It is my understanding a facilities based CLEC is a FCC designation. There are
rules that govern who is considered a CLEC. We are a VoIP based interconnect
carrier based on the rules. Even though we offer internet services. We do not
own any phone service exchanges.
If you can operate out
I know this is probably a very basic question for many on this list. But in
troubleshooting an issue, I wanted to take a step back ask the question. In
Asterisk (or maybe all SIP), how do extensions stay registered with the SIP
server?
Do the extensions simply register repeatedly as a means
SIP normally doesn't use TCP, it uses UDP, and is sessionless in that
context. The exact mechanics of a registration can get deeply involved, so
I'm going to give a very cursory overview. The endpoint tells the server
(Asterisk, or whatever) that it would like to register, with a username and
Extensions do not register - peers do. A peer can register itself or be
registered by Asterisk. In most cases the extension is equivalent to the
peer (301 = 301) but it can be quite different (301 = sipuser1) or (301 =
d...@impalanetworks.com).
From: asterisk-users-boun...@lists.digium.com
I think the question is more along the lines of how does asterisk know
immediately when a sip phone becomes on line and when you
unplug the phone from the network, how does asterisk essentially know
immediately that it status is UNKNOWN
If I am not mistaken.
--E
From:
Just trying to offer a little enlightenment - There are basically two
methods of sip phone (peer/extension) registration. Method 1 is
self-registration where Asterisk does not know or care about the phone
until it asks to register. Method 2 is required-registration where
Asterisk expects the
The SIP server has no way to tell the device is no longer available until the
next time the device registers (or the server tries to send a call to the
device).
ASTERISK has the qualify feature, which uses a SIP OPTIONS packet to probe the
peer every min or so.
-Original Message-
I think the registration part was answered. The de-registration part is
different. If the phone is gracefully taken off line it specifically
de-registers. If it just can't be reached because it powers off or the
router closes NAT, or whatever, then Asterisk won't know this until it
times out.
I think the wrap up answer is the interval of registration compacted, if used,
with the SIP OPTION packet.
I like the SIP OPTION packet because we have scripts to monitor the status and
lets us know when a phone is up or down.
--E
From: asterisk-users-boun...@lists.digium.com
Un-top-posting and de-crufting...
On Mon, Nov 14, 2011 at 3:19 AM, Tristram Cheer wrote:
I'm using DumpChan(1001)...
I would like to dump this output to a file specifically for DumpChan...
On Mon, 14 Nov 2011, Warren Selby wrote:
If you call DumpChan from an AGI you should be able to
I have a home asterisk box which connects to the office asterisk, so I
can just dial extensions.
This used to work just fine. I'm using 10.0-rc1 on the home box, 1.8.7.0
on the office. But it doesn't work now:
[Nov 14 18:38:19] NOTICE[21563]: chan_sip.c:13161 sip_reg_timeout:--
You're not going to get a telnet connection on port 5060, since that's tcp and
sip uses UDP.
Use tcpdump/wireshark on your office pbx to see if the packets are getting to
you. If not, then there's something wrong inbetween.
A firewall misconfig, perhaps. Or the unthinkable: your home ISP
On Mon, 14 Nov 2011, sean darcy wrote:
telnet localhost 5060
Trying 127.0.0.1...
telnet: connect to address 127.0.0.1: Connection refused
Telnet is TCP while SIP is usually UDP.
The 'Connection refused' just means you don't have telnetd running (a good
thing) or anything else (xinetd,
I think that you actually should be looking to your state. I'm pretty sure that
even if CLEC is an FCC designation, it is implemented either on a per-state or
per-LATA basis. Here in NM there's only 1 LATA, which is why I'm not completely
sure. But I believe that the CLEC qualifications
On 11/14/2011 08:33 PM, Alex Balashov wrote:
There is no free lunch. There is no such thing as an easy-peasy
regulatory reclassification that gets you the same stuff you were
paying before, but more cheaply.
*paying for before
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree
Worst reason to become a CLEC: improved cost structure. Or, to be
precise, it is a counterfactual reason, because it does not result in
improved cost structure.
This idea is driven by an incomplete understanding of what being a
CLEC entails, or, for the less critically thoughtful, the free
Agreed. And facilities based CLEC even scarier.
Regulatory / billing / PUC legals etc ugh
Sent from my iPhone 4S
On Nov 14, 2011, at 8:33 PM, Alex Balashov abalas...@evaristesys.com wrote:
Worst reason to become a CLEC: improved cost structure. Or, to be precise,
it is a counterfactual
Hi all，
Recently，I met a very strange phenomenon。I found that my asterisk bin file
had changed when running。I checked a lot of machines ， and the result is almost
all of the bin files have taken place。
the following is the result of the calculation。
[root@callcenter beijin]# ls -l
On 11/14/2011 07:56 PM, Douglas Mortensen wrote:
I think that you actually should be looking to your state. I’m
pretty sure that even if CLEC is an FCC designation, it is
implemented either on a per-state or per-LATA basis. Here in NM
there’s only 1 LATA, which is why I’m not completely sure.
The ride is over before it even began A local ILEC here in Canada,
is already offering
Unlimited World service. And this on a Tier 1 network, not the crap
we're use to doing
business on. Choose a different angle before you get anymore grey
hairs on that head...
On 11/14/2011 08:36 PM, Robert-IPhone wrote:
Agreed. And facilities based CLEC even scarier.
I'm curious what sort of thing would be considered a non-facilities
based CLEC, since UNE-P was cancelled in 2003.
There are some non-interconnected CLECs out there that exist for the
sole purpose
Wow so I left before the end of resale Verizon UNE then.
We ran Lucent 5E and Nortel DMS and provided facilities voice and DSL.
Having a large SONET fibre infrastructure helped too.
Sent from my iPhone 4S
On Nov 14, 2011, at 8:53 PM, Alex Balashov abalas...@evaristesys.com wrote:
On
UNE is alive and well. UNE-P is what's gone.
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity, errors, and general sloppiness.
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax:
On Mon, 2011-11-14 at 20:51 -0500, Nick Khamis wrote:
The ride is over before it even began A local ILEC here in Canada,
is already offering
Unlimited World service. And this on a Tier 1 network, not the crap
we're use to doing
business on. Choose a different angle before you get anymore
Hahah! Yeah it does doesn't it? What do we do? How do we stay
a float, It almost seems like the ILECs will drop their rates to a
penny once the people in this, and Kamailio lists ;) actually put a
dent in their underline.
Nick
On Mon, Nov 14, 2011 at 9:08 PM, Jeff LaCoursiere j...@sunfone.com
Only through new, innovative applications. They will always deliver transport
and dialtone cheaper than you.
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity, errors, and general sloppiness.
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street
On 11/15/2011 02:45 AM, jordan pan wrote:
Recently，I met a very strange phenomenon。I found that my asterisk bin
file had changed when running。I checked a lot of machines ， and the
result is almost all of the bin files have taken place。
[snip]
Maybe prelink? See man prelink.
Regards,
Patrick
On 11/14/2011 07:05 PM, James Sharp wrote:
You're not going to get a telnet connection on port 5060, since that's tcp and
sip uses UDP.
Use tcpdump/wireshark on your office pbx to see if the packets are getting to
you. If not, then there's something wrong inbetween.
A firewall misconfig,
Yeah! That is what I was thinking... Bringing Voice and Video under
one umbrella, things like that...
I actually come from a speech recognition and natural language
processing background. Trying to
build the voice network, and seeing how I can bring it all together.
P.S. I started by getting
On 11/14/2011 09:57 PM, sean darcy wrote:
Unthinkable!! Used wireshark: I can see the REGISTER packets going out
from the home router, but nothing from home:5060 shows up at the office.
Bummer. Now I get to think about how to set up special ports between
home and office. A great evening
Continuing eherr here, behind the OPTIONS messages(infact all SIP comm) you
definitely to look into SIP timers which tell how many time to resend a
packet if no response is received and for how long to wait before thinking
that the SIP packet got lost(network disconnected or end-point lost)
so,
There are clever ways to be a CLEC, and keen reasons for becoming so. But
cheaper stuff ain't one of them.
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity, errors, and general sloppiness.
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street
Hello,
I have a testing scenario at hand. I want to make a call from Asterisk
CLI or AMI to an external network gateway. Is this possible.
Let me explain the use case.
Asterisk server (say 192.168.5.10) has few registered endpoints or softphone.
Now an external gateway (say my-example.com or
Hey,
Though your requirements are unclear and below may not exactly fit your
specs unless you give some more usage details.
if your gateway requires no authentication, yes you can do this by writing
a dialplan extension like below
exten = calling-togw,1,NOOP(I'll be getting some variables from
Amar,
In general, gateways don't register. They are simply defined as a peer and
calls are routed to them in the dialplan. When I do this I usually use the
local channel to get to the
dialing contexts.
Get in touch if you need a more detailed example
Bruce Ferrell
On 11/14/2011 10:01 PM,
46 matches
Mail list logo
