2009/7/14 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
Just been contacted by a UK Enum registrar looking for ITSPs to become
resellers of their Enum registration systems ...
Is anyone using Enum?
Does anyone (other than cynical old me) think that Enum is a
Inspired by AG Projects' Adrian Georgescu's post of Eric S. Raymond's
classic How to Ask Questions the Smart Way to the OpenSIPS-users
mailing list[1], I'm going to repost it here:
http://www.catb.org/~esr/faqs/smart-questions.html
As Adrian said, This a good read for those who show up on
Yes, provided they are inclined to look at your issue. Do not be
presumptuous or self-entitled.
Zeeshan Zakaria wrote:
It should be an easy one for many of the experts here.
On Mon, Jul 13, 2009 at 8:10 PM, Zeeshan Zakaria zisha...@gmail.com
mailto:zisha...@gmail.com wrote:
For a
Oh, I forgot to mention, the customer can make outbound calls from this
extension. Just calls cannot be routed back even though the IP Address
and port are in the realtime cache.
Ishfaq Malik wrote:
Hi There
I have an extension which is in a different country and is constantly
lagged
800ms is horrendous lag for a VoIP connection. If I were you, I'd be
investing some time in finding out why the lag is so great. Even if I
do a ping to a UK address, I'm getting pings of no more than 300ms from
Australia. Unless you've got multiple satellite connections in the path
(in which
Cheers Rob, I was thinking it was due to a low bandwidth connection at
the other end but from what you're saying it sounds like this is not the
case.
Rob Hillis wrote:
800ms is horrendous lag for a VoIP connection. If I were you, I'd be
investing some time in finding out why the lag is so
I'm tearing (what's left of) my hair out on this one :-(
shortform
How can I set the CDR(userfield) in the calling thread from the dialplan
(actually a macro called from a feature) in the called thread?
long version
I use mixmonitor to record calls driven by entries in the asterisk
database
On Wed, Jul 15, 2009 at 02:19:20PM +0300, Gad Alaloof wrote:
Hi
I'm new developer on Asterisk and i have some questions:
I suspect you ended up on the wrong list.
Generally this is the list for those who mess with the actual C code of
Asterisk. I suggest you follow-up on asterisk-users (CC-ed
Alex, no need for personal attacks.
Zeeshan
On Wed, Jul 15, 2009 at 2:23 AM, Alex Balashov abalas...@evaristesys.comwrote:
Yes, provided they are inclined to look at your issue. Do not be
presumptuous or self-entitled.
Zeeshan Zakaria wrote:
It should be an easy one for many of the
Low bandwidth is another possibility, but I'd have though that any
connection slow enough to generate that much latency wouldn't be usable
for VoIP in the first place.
Ishfaq Malik wrote:
Cheers Rob, I was thinking it was due to a low bandwidth connection at
the other end but from what you're
Phillip, thanks for your advise. I tried that, but it didn't help.
I'll keep experimenting with it, and will post solution if found one.
Zeeshan
On Wed, Jul 15, 2009 at 8:20 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
Alex, no need for personal attacks.
Zeeshan
On Wed, Jul 15, 2009 at
Hi there,
I'm new to this list, so sorry if this has already been asked before
[hopefully this is the correct list too!].
I'm in the process of building my asterisk system. I've set up a CentOS
machine, together with Cisco 7912 phone over SIP. I can make and receive
calls from outside over my
It is not a personal attack. It is a statement about the appearance
that your manner of formulating your follow-up creates, intentionally
or otherwise.
Experts do not like being prodded; and take a critical view of
rhetorical manipulation techniques designed to appeal to their baser
A useful guide; Sadly many posters won't read any of it. We have all kinds
here, but the overall experience can be quite good and useful if you apply
patience and principles.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Understood.
Zeeshan
On Wed, Jul 15, 2009 at 8:50 AM, Alex Balashov abalas...@evaristesys.comwrote:
It is not a personal attack. It is a statement about the appearance that
your manner of formulating your follow-up creates, intentionally or
otherwise.
Experts do not like being prodded; and
In my shop, we got a better router to support QOS and configured our Polycom
phones to always request highest levels (UDP gets 6, everything else gets
3).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
Hi all,
I want to try to use a USB phone with Ekiga under Linux (Debian Lenny). It
works: I can receive and make calls. But some buttons of USB phone don't
work properly. In particular, button *, #, and hangup have wrong key
mapping.
Someone have tried a USB phone
Thamks all
Marco
As a matter of personal opinion, I think 90% of the useful takeaways
for this specific mailing list don't have so much to do with politics
and attitude as with asking specific questions of a manageable scope
and formulated in an addressable way.
Many questions lack conceptual integrity and
You don't say what Technology you're using to connect the phone to Asterisk
or what release of Asterisk you're working with. I know that in 1.4SVN, the
* and # are sometimes non-respondent on incoming DAHDI calls (can't use
features because can't do *1, #1, etc.). More information would help.
On Wed, 15 Jul 2009, Danny Nicholas wrote:
In my shop, we got a better router to support QOS and configured our Polycom
phones to always request highest levels (UDP gets 6, everything else gets
3).
Did this apply to your connection to the net, or just internally? I am
most concerned with
Ours is just internal, but the concept should be the same. My boss could
talk on his phone fine until he cranked up Foxnews feed. Once the video
started, he couldn't talk on his phone anymore (bad quality or total loss of
call).
-Original Message-
From:
Hi all,
We we send a server out to a customer with a PRI card I basically
tell the PBX person that we want national, NI2, b8zs, D channel is on 24,
and we are the CPE and your the NET and us a cross over cable.
Allow extensions, local calls and long distance.
on the surface I would have thought
(Both on Asterisk 1.2 and 1.4)
I was struggling to find out why my CDR was recording dst = h after a call
hangup. It was working fine until I added a GotoIf statement before ResetCDR
to calculate some value for userfield column. Today I tested and found out
that if ResetCDR is put after GotoIf
Danny Nicholas wrote:
Ours is just internal, but the concept should be the same. My boss could
talk on his phone fine until he cranked up Foxnews feed.
Therein lies the problem
One should NOT contaminate their network with Fixed News ( AKA as Fox
Noise )
In addition to overloading internal
On Wed, 15 Jul 2009, Marco Sambo wrote:
Hi all,
I want to try to use a USB phone with Ekiga under Linux (Debian Lenny). It
works: I can receive and make calls. But some buttons of USB phone don't
work properly. In particular, button *, #, and hangup have wrong key
mapping.
Someone have
On Wed, 15 Jul 2009, Danny Nicholas wrote:
You don't say what Technology you're using to connect the phone to Asterisk
or what release of Asterisk you're working with. I know that in 1.4SVN, the
* and # are sometimes non-respondent on incoming DAHDI calls (can't use
features because can't do
I've found a work around, i.e. if put the dialout command in a separate
macro, then CDR records the values fine, whether the hangup macro is called
by the original context, or by the dialing macro.
Posting here in case somebody facing a similar issue like me can benefit
from it:
[test]
exten =
Hi Gang,
Running Asterisk 1.4SVN using Polycom 501 phones. Just enabled
CallerID and for the most part it works as good as you'd expect anything to
from the phone company to. Except: on about 1 out of 10 transfers, instead
of getting a callerid of joe cool 100 or abc company
To do the same in AEL, use 'catch' construct (ref
https://issues.asterisk.org/view.php?id=14956). This is so that we can catch
extension h and return to the calling macro, so that it can continue its
priority after the Dail command. I've tested it and it works.
context test {
_NXXNXX = {
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Danny Nicholas wrote:
Hi Gang,
Running Asterisk 1.4SVN using Polycom 501 phones. Just
enabled CallerID and for the most part it works as good as you’d expect
anything to from the phone company to. Except: on about 1 out of 10
That's the bizarre thing; I get the ID from telco and whether it passes on
to the extension depends on how the transfer is done.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L. Kline
Sent: Wednesday,
On Wed, 2009-07-15 at 08:10 -0500, Danny Nicholas wrote:
In my shop, we got a better router to support QOS and configured our Polycom
phones to always request highest levels (UDP gets 6, everything else gets
3).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Wed, Jul 15, 2009 at 9:40 AM, Jerry Geisge...@pagestation.com wrote:
Anyway as example. the last customer I told the above information. He
set up the PBX
and I can make 4 digit calls successfully, 7 digit and long distance are
not successful.
I don't know what's going wrong, but the
That's not bizarre at all. Blind transfers will always forward the other end's
CID. Attended transfers will always forward the CID of the phone doing it.
Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP
- Danny Nicholas da...@debsinc.com
On Mon, Jul 13, 2009 at 8:10 PM, Zeeshan Zakariazisha...@gmail.com wrote:
Any idea why is this happening and how can I have correct 'dst' value if the
caller hangs up first.
[dialout]
exten = _NXXNXX,s,1,Dial(SIP/XX/${EXTEN},30)
What happens when you put a
exten =
Beg to differ. Call comes in from telco as 201212. If I do a blind
transfer, phone shows asterisk. If I do attended transfer, ID shows as
201212. I'm not saying thats how it's supposed to work, just how it
does in my shop.
-Original Message-
From:
David, to answer your question, if I put NoOp(${CDR(... in the same context,
it shows the correct destination.
Dealing with CDR values have been a pain, and I know this because I've made
two billing systems in last two years. For AGI, there is DeadAGI, which
helps, but this current scenario was
Hello!
I set my devices to only use g77a, but I am getting this when I run
show channel
NativeFormats: 0x8 (alaw)
WriteFormat: 0x8 (alaw)
ReadFormat: 0x4 (ulaw)
Why is ulaw (g711u) showing up for the Read Format?
Thanks,
Elliot
___
--
Hello,
The call center I manage previously had almost all calls entering a
single queue. In order to differentiate the calls to the techs we set
the callerid name based on the caller id number offered to us.
Basically, it was a gosubif the callerid number matches this, and in
the sub we set the
On Tue, 14 Jul 2009 22:58:14 + (UTC), Jeff LaCoursiere wrote:
Jeff, yeah i saw the posts, i followed Bob Pierce config and had no
luck, BUT it just started to work, i changed AP's, seems like theres
something wrong with Ubiquiti NanoStation2 WMM implementation, i used a
Linksys WRT54G2
Hi all,
Just a quickie to say that this has been solved now - real simple -
downloaded '*current*' rather than the versions from the home page of
Astrisk.org. (didn't realise there was a 'current' version tbh.
Anyways - I don't get Asterisk seg faulting now when hammering the
speaker button on
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I must be missing something here but I can't figure out why I can't get
DEVICE_STATE() to give me anything other than NOT_INUSE.
I have two extensions: and 6668. I used 6668 to make a call to
yet another phone, so I know that it's busy. I
Barry L. Kline wrote:
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I must be missing something here but I can't figure out why I can't get
DEVICE_STATE() to give me anything other than NOT_INUSE.
I have two extensions: and 6668. I used 6668 to make a call to
yet another phone, so
Barry L. Kline schrieb:
I must be missing something here but I can't figure out why I can't get
DEVICE_STATE() to give me anything other than NOT_INUSE.
I have two extensions: and 6668. I used 6668 to make a call to
yet another phone, so I know that it's busy. I then use to call
Thank you all for your input into this question. It is very helpful to get
your opinion and experience with this matter.
I mean in my case a single server application. And what I'm probably going
to have to do is use AMI via either a windows .net application that will
parse and monitor the ami
I am having trouble with a DID on a PRI. If there is a call to that DID (let
say 5551234) , the next calls get a busy signal. How to I go about sending
the call to the next available channel ?
Thanks!
G.
___
-- Bandwidth and Colocation Provided by
I would like to setup an iphone to be an extension on my pbx. I have looked
at SIAX as well as Asteriskc2d. Does anyone have any experience with either
of these or another app? The important thing for me is that I can run it in
the background so I can always be available to receive a call. It
Gondar Monn wrote:
I am having trouble with a DID on a PRI. If there is a call to
that DID (let say 5551234) , the next calls get a busy signal. How to
I go about sending the call to the next available channel ?
Thanks!
G.
If the telco is providing the PRI then you need to tell
As far as I know this can't be done. I have configured my iphone 2G, using
WeePhone, as an extension on my Trixbox. However, the WeePhone does not run
in the background. Apple has a pretty strong block on doing this with any
app.
David
_
From:
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Mark Michelson wrote:
You need to set a call-limit for the SIP peer. Device state calculation for a
SIP peer is predicated on both the call-limit and busylevel. Let's say that
you
were to have a call-limit of 2, but no busylevel set. These
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Philipp Kempgen wrote:
Just to be sure: Do you have hints configured for the extensions?
See http://das-asterisk-buch.de/2.1/blf-leds.html
(The text is in german but there are many examples in extensions.conf
and extensions.ael syntax. Zurück =
Rollover or hunting is generally the default on PRIs. It sounds like
Gondar's concern is with a specific DID number (Do multiple calls to other
DID numbers work OK?). I'd wonder about a couple things:
Are people dialing '5551234' directly, or are calls being forwarded to that
number? Some
Thank you for your quick answers!
@ Brent: rollover is on, I would like to any calls that come on 5551234 to
another DID, to be able to receive several calls on the same number
@ Don: You are right, I am talking about a specific DID: We have an analog
line with busy forward setup @ the telco to
Forwarding a POTS line will not work, it is like a trunk to trunk transfer
so it is not free, so the line stays busy.
You need to port that number over to the PRI provider.
On Wed, Jul 15, 2009 at 7:41 PM, Gondar Monn gonda...@gmail.com wrote:
Thank you for your quick answers!
@ Brent:
It's quite possible that your busy forward will only forward one call at a
time. What happens if you dial multiple calls directly to 555-2345? If that
works, the problem is not with your PRI and Asterisk, the problem is with
the forwarding from 555-1234.
--Don
_
From:
Steve Totaro wrote:
Forwarding a POTS line will not work, it is like a trunk to trunk
transfer so it is not free, so the line stays busy.
You need to port that number over to the PRI provider.
That all depends on the POTS provider.
Multiple calls from one POTS number CAN be done, but
There's an application called Voipover3G. You have to jailbreak your iphone
first. Also, there's backgrounder.
Jimmy
-Original Message-
From: da...@slopecolorado.com
Sent: Wed, 15 Jul 2009 16:36:54 -0600
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Iphone setup
Always a great readthanks.
PaulH
Alex Balashov wrote:
Inspired by AG Projects' Adrian Georgescu's post of Eric S. Raymond's
classic How to Ask Questions the Smart Way to the OpenSIPS-users
mailing list[1], I'm going to repost it here:
Un-top-posting...
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
Balashov Sent: Wednesday, July 15, 2009 1:19 AM
Inspired by AG Projects' Adrian Georgescu's post of Eric S. Raymond's
classic How to Ask Questions the Smart Way to the OpenSIPS-users
mailing list[1], I'm
Hello,
this is what I'm trying to accomplish:
- receiving an inbound call from A
- dialing another number (B)
- bridge A and B
- every x minutes, debridge A and B, and bridge A with C (SIP call to an
platform that is gonna play an ad)
- rebridge A and B
Any advice on how to do this?
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