2012/1/12 Ishfaq Malik i...@pack-net.co.uk
Hi
I'm using 1.8.7.0 with the RealTime architecture.
If a call goes into application Queue and is abandoned by the caller, no
entry is made in the CDR. Entries are made into the queue log.
This cannot be correct behaviour, all calls should show
On 10/14/10 9:10 PM, Jeff LaCoursiere wrote:
Hi,
Embarrassed as I am to write this, I am hoping for some advice. One of
our very first PBX installs, now six years old, was taken advantage of
over the past few weeks. A victim of sipvicious, I assume, that managed
to guess one of the SIP
On 8/20/10 1:24 PM, A J Stiles wrote:
On Wednesday 11 Aug 2010, Tilghman Lesher wrote:
On Wednesday 11 August 2010 03:59:28 A J Stiles wrote:
I'm having a problem with Asterisk 1.6.2.9 with the MySQL cdr addon.
With some calls, the value in the `billsec` field in the CDR is exceeding
the
Supposedly, the filters drop it in the transaction stage. But for some
reason, every time I get dropped from the list, it's just after a spam
email was sent out en masse, so I'm not sure what's up there.
On 7/28/10 10:43 PM, jon pounder wrote:
SIP wrote:
what can you do ? simple discard
On 7/28/10 9:45 PM, Sam wrote:
Just a note, the asterisk mailing list server continually gets
blacklisted over at
http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering
mail to spamtraps. Perhaps something needs to be looked into...
Regards,
Sam
Spammers sign up to the
On 6/23/10 7:20 AM, RSCL Mumbai wrote:
Hi,
Looking for some reliable and quality providers of USA DIDs.
Any pointers ?
Thx
Sans
We've had good luck with Vitelity and DIDForSale.com.
N.
--
_
-- Bandwidth and Colocation
Danny Nicholas wrote:
Also cheaper to replace flash card than hard drive.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Monday, June 14, 2010 4:21 PM
To:
fan
* Hard drive (not flash memory)
Capabilities/capacity
* No GUI, no X
* Register to multiple SIP servers
* There will be no PSTN
* No analog phones
* Small number of SIP devices will register - maybe 10 max
* Three simultaneous channels active
* Skype
Jeff LaCoursiere wrote:
On Thu, 20 May 2010, Gordon Henderson wrote:
On Thu, 20 May 2010, SIP wrote:
Even IF you could get a keyboard with lights you could individually turn
on and off (never seen one),
http://www.artlebedev.com/everything/optimus/
Bit expensive though
Tzafrir Cohen wrote:
On Wed, May 19, 2010 at 07:46:26AM +0200, Olivier wrote:
2010/5/18 Danny Nicholas da...@debsinc.com
Dumb question – wouldn’t it be easier to monitor a web interface than a
phone with 100 lights?
Yes and no : operator already has a Flash Operator Panel
On 5/2/2010 4:52 PM, Steve Edwards wrote:
On Sat, 1 May 2010, SIP wrote:
[snip]
We run DeadAGI for a considerable number of calls since it has the
ability to run post-hangup cleanup no matter which side hangs up (unlike
AGI).
[snip]
When a channel hangs up, Asterisk sends
On 4/30/2010 6:03 PM, Luki wrote:
It is irrelevant who hangs up, you want to just use DeadAGI in the h
extension
I wish that would be the case, but at least on 1.4 I see:
[Apr 30 14:59:38] -- Executing [...@master-route:1] DeadAGI(...) in new
stack
[Apr 30 14:59:38]
What country are you in? Makes somewhat of a difference.
N.
On 3/17/2010 8:49 PM, Mike wrote:
Ok, I see there's alot out there of voip providers.
Curious what to watch out for ? charges and fee's, etc ?
If anyone has feedback as to a GOOD voip provider experience (one that
gave FREE DID)
Well. For a free US DID, you could check IPKall (http://www.ipkall.com/)
I've no idea about quality, since it's been almost five years since I've
even LOOKED at them. But then generally work well with Asterisk. And
they're free.
N.
On 3/17/2010 9:09 PM, Mike wrote:
My bad, I'm in Los angeles
Will Payne wrote:
On 8 Mar 2010, at 22:08, Dave Poirier wrote:
Top posting to remain consistent...
I drop litter because everyone else does.
;)
W
Different entirely. People who switch to bottom posting on a top-posted
thread make things MUCH harder to read by being
Will Payne wrote:
it just seemed like a 'I know this is wrong, but...' comment :)
Quoting entire emails is bad, m'kay. Quoting whole threads is worse. If you
snip the quote down to the relevant portion, you can reply where you like,
regardless of what's gone on beforehand.
(Surely there's
David @ULC wrote:
I use IdeaSip with IPKall.
How may channels are open when we use IdeaSip ?
Incoming IdeaSIP SIP channels are unlimited; however, I believe IPKall
limits you to 94 channels via their DIDs.
You would, of course, need the bandwidth to be able to handle 94
simultaneous
Julian Lyndon-Smith wrote:
Has anyone managed to get these two phones to make a video call to each other
?
If so, care to share how the hell you managed ?
the GXV is at the latest firmware, and xlite the latest download
Asterisk 1.4 trunk
TIA
Julian
Yes. Have done it often.
Philipp Kempgen wrote:
Leif Neland schrieb:
Norbert Zawodsky skrev:
The number +43-1-3207978 is my telephone number. I own it as long as I
pay for it. And with extra digits behind it I can do whatever I like. I
can create any extension - physical or virtual. I can attach a
Benny Amorsen wrote:
SIP s...@arcdiv.com writes:
It may work in Austria, and may even be valid in Austria. But if that's
the case, it's because Austrian dialing is a complete hack -- NOT
because that's the way it's intended OR designed.
Err no? It's perfectly sane
Raimund Sacherer wrote:
Adding random digits to a PSTN and expecting to get the same person at a
different extension you don't think that's a hack? I do. One should
Sorry, please do not call a whole country using a hack when their solution is
legitimate.
Austrian PSTN
John Novack wrote:
Raimund Sacherer wrote:
Adding random digits to a PSTN and expecting to get the same person at a
different extension you don't think that's a hack? I do. One should
Sorry, please do not call a whole country using a hack when their solution is
exist as pstn ?
A simple example:
My pstn number is +43-1-1234567. Everybody around the world can call
me using this number.
Lets say, I have 3 extensions: 0=reception, 10=secretary, 20=boss.
If someone calls
ENUMLOOKUP(+4311234567) he will get a uri sip:0...@ip.of.my.asterisk
ENUMLOOKUP
and dig
7.6.5.4.3.2.1.1.3.4.e164.arpa both return NAPTR records.
Now for the less clearer points:
Your'e supposed to register your number without any extension.
If I have some extensions here, how can the calling party get the
correct sip uri to the requested extension
Does the phone have some sort of NAT Keepalive setting? Often, the only
way to keep that port open on the user's NAT gateway is to have the
NATted client send the occasional data out through the port.
N.
Ron wrote:
i have also tried setting qualify='yes' but cpu usage spiked to 100%.
Ron
...@lists.digium.com] On Behalf Of Lee Howard
Sent: Thursday, November 12, 2009 12:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] allowguest defaults to yes for SIP
Tilghman Lesher wrote:
The issue in question was suspended, while the reporter
Remco Barendse wrote:
On Fri, 30 Oct 2009, hbk wrote:
Hi,
I have played with the 820 for some weeks, mostly love it excellent speech
quality. Even got the mini browser running
showing my favorite webcam, this could be put to real use too:)
Issues so far:
Some embarrassing crashes
Sounds like it wasn't a very interesting track. ;)
N.
Danny Nicholas wrote:
Is THAT a summary :)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randy R
Sent: Wednesday, October 21, 2009 1:24 PM
To:
Kevin P. Fleming wrote:
SIP wrote:
In an ideal world, when Asterisk sent an ACK, whatever server/client it
was connected to would respond accordingly. It is, however, not an ideal
world, so this doesn't always happen.
This is not correct; there are no responses to SIP ACK messages
Alex Balashov wrote:
SIP wrote:
What is your citation for this qualification? RFC 3261 does not seem to
me to say that, as in 13.1:
Because of the protracted amount of time it can take to receive final
responses to INVITE, the reliability mechanisms for INVITE
Gianni Fioretta wrote:
Hello.
I have a problem with Asterisk, sometimes it hangs up an external call after
20 seconds, apparently without any reason.
The call comes from a SIP server hosted from EuteliaVoIP, many peers rangs
and one of them answer, the call ends itself after 20 seconds
and configure, and
comparable in price. Very SIP/VoIP friendly. Loads of optional modules
(we use its mail filter module to filter spam/viruses for several
hundred thousand user mailboxes, for instance) to limit the cost to what
you need.
Also has a built in SIP Proxy, although I've never used
-6001 (London in-dial).
I don't know if I'll be able to make the call but my guess is he's
referring to the SIP trapezoid:
http://www.iptel.org/sip_trapezoid
The SIP trapezoid is a concept/teaching tool usually used in
situations involving a proxy (or multiple proxies) illustrating
An Asterisk MeetMe conference sounds like the ideal sort of scenario for
you, allowing people to join in or drop off during a session as they
please.
N.
li...@mgreg.com wrote:
Hi All,
As is obvious by my joining the list, I'm interested in learning more
about Asterisk. I have
When you start taking credit card payments (assuming you will), be
careful with small payment amounts. You'll become a fraud haven. A lot
of CC thieves or people who've just bought a CC number will use a small
amount charge to check and see if the card is any good.
Check out some of the MaxMind
A quick look at the system shows you're not logged in, which is why
you're getting that message.
N.
David @ULC wrote:
Oh my god..
Today its saying there is NOONE to take your call.I am using IdeaSIP
What could be the reasons ?
It was working perfectly till saturday .
On Thu, Aug 20,
Means your username is not registered on the IdeaSIP system (your
client/phone is not logged into IdeaSIP).
N.
David @ULC wrote:
you're not logged in means ?
On Mon, Aug 24, 2009 at 11:39 PM, David @ULC ucoms2...@gmail.com
mailto:ucoms2...@gmail.com wrote:
Oh my god..
IdeaSIP, GizmoProject, IPTel, maybe OnSIP (don't quote me on that one,
I'm not sure, but someone around has surely used it), etc, etc. There
are a lot of alternatives about.
Disclaimer: IdeaSIP is my personal unruly child (hence top billing on
the list of alternatives).
N.
David @ULC wrote:
SIGNED MESSAGE-
Hash: SHA1
SIP wrote:
Daniel,
Hi SIP.
Check your stunaddr setting. Is it misspelled, or do they really use
stun.exiga.net instead of stun.ekiga.net ?
Thanks to indicate that error to me. I doing the test again. I don't
believe that this solves
/wiki/view/OpenPBX.org+FAQ;):
The main points listed on Asterisk's CONS that concerned me were:
* Conferencing on Asterisk depends on Zaptel hardware and/or kernel
modules for timing;
* Lack of built-in STUN support for SIP NAT traversal;
* Asterisk doesn't use
to expect.
You mean comforts which you have come to expect. Again, my needs have
been so far fulfilled for conferencing and SIP/PSTN gateway uses.
Pointing to particular missing applications instead of making your own
analogy would be useful, otherwise you are not really being of much
help
/asterisk/extensions.conf:
[from-internal]
...
exten = _8.,1,Dial(SIP/ekiga/${EXTEN:1},20,r))
I tried a echo test, dialing in my case to 8500, but in spite of seeing
traffic towards Internet, nothing is heard. Setting 'sip set debug', I get
the following thing:
--- SIP read from 10.1.0.65
Tony Mountifield wrote:
In article 05d03313-994b-4892-b045-f61332ddb...@geekinter.net,
Steve Howes st...@geekinter.net wrote:
On 14 Aug 2009, at 09:17, Neeraj Chand wrote:
Asterisk version 1.4
From: Neeraj Chand
Sent: Friday, 14 August 2009 8:17 PM
To:
I'm not sure there IS an issue, per se. There are lower bitrate codecs
that will work fine for voice communications in both directions. But if
you're trying to force a low-end codec to the upstream, that just means
the downstream on the remote end is going to be stuck with a low-end
codec. And if
I've had similar results to you. Packet loss even when not transcoding.
Overall poor performance across the board. We considered it a failed
experiment.
N.
Zoa wrote:
I have played with DD-WRT on linksys wrt54g version 5 last week (2
different ones, they are the model with less memory so
Alex Balashov wrote:
BTW, if you need a generic, scalable, high-volume B2BUA, it is not a
best practice to use Asterisk for that purpose.
Indeed. But you can grow some good SMB B2BUA systems out of it.
Freeswitch would be a grand alternative... if it had documentation.
Anywhere. Ever.
Dunc wrote:
Doug Lytle wrote:
Your membership in the mailing list asterisk-users has been disabled
due to excessive bounces The last bounce received from you was dated
Anybody else seeing this? My mail server logs don't show any issues.
Doug
I just did yes,
Michael wrote:
On Tue, 14 Apr 2009 20:47:29 you wrote:
Hi michael,
you should open both tcp,udp 5060,5061 too and as you mentioned between
1-2.
AFAIK 5061 TCP is for TLS SIP which isn't used much yet?
Is TCP the default for 5060, with UDP as fallback, or is this provider
, if I am using * to talk to a cisco
gateway via SIP, is there some sort of encryption you can use? Like a
vpn tunnel?
Can someone capture packets and re-assemble to make out a conversation?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users
IPKall still exists.
http://www.ipkall.com
No customer service, and the number has to be used every month or you
lose it. But it's there. And free. And good.
N.
Dean Collins wrote:
Does IPKALL still exist?
I am after a free SIP trunk – who is still giving these away these
days? As I
Daniel Nowacki wrote:
SIP wrote:
IPKall still exists.
http://www.ipkall.com
No customer service, and the number has to be used every month or you
lose it. But it's there. And free. And good.
I get an ugly 404 when trying to sign up or log in... That is probably
abandonware
randulo wrote:
This brings up a side issue. Banks on the Internet have had to provide
a sort of insurance that allows the customer to be protected if
someone hacks in to his or her account. ITSP will need to think
carefully about having a similar policy that protects people from an
attack to
randulo wrote:
On Thu, Mar 26, 2009 at 1:32 PM, SIP s...@arcdiv.com wrote:
As an end-point ITSP, I can assure you, it would be us who's assessed
the requisite charges. If someone uses a fraudulent card, we're required
to pay. If someone uses a three letter password on his account, and it's
randulo wrote:
On Thu, Mar 26, 2009 at 2:38 PM, SIP s...@arcdiv.com wrote:
And so, in answer to your question, I don't think there ARE necessarily
steps that can be taken right now to ensure that there's a rational
approach to the resolution of such an issue of fraud. Barring some sort
randulo wrote:
On Thu, Mar 26, 2009 at 4:19 PM, SIP s...@arcdiv.com wrote:
The first approach is the current approach: build software with little
thought to how it will be secured, opting for all the work of securing
What about SIP itself? Does it provide enough crypto to be solid
Dave Platt wrote:
SIP was written in such a way that the hashes it sends for passwords
could, with only a trivial rewrite of the server code, be SHA1 instead
of MD5 -- which would increase security to the level that, currently, it
would be far more trouble than it's worth to even bother
David Ruggles wrote:
I was looking at the aastra 9133i, however I was informed that this phone is
no longer supported. What are good phones around the $100 - $125 price
point? (Need POE)
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200
Not to burst your bubble, Jon, as I agree with a majority of what you
said... but using an argument about the evolution of email to support an
argument about how telcos should have better tracking and accountability
is somewhat weird.
We get 3 million email messages a day through our servers.
,
and replaces it with something that's clearly just a kludgy add-on to a
product which was never originally designed for the task.
One thing that bothers me with the current crop of hardware SIP phones
is that they are hopelessly properitary.
So what would it take to build a fully-adaptable
Michael wrote:
This has absolutely nothing to do with the fact that something is
opensource. The fact that the source is open has nothing todo with its
pricetag. Sometimes opensource products are more expensive then closed
source products.
If you want
Grygoriy Dobrovolskyy wrote:
2009/2/13 Tzafrir Cohen tzafrir.co...@xorcom.com
mailto:tzafrir.co...@xorcom.com
On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote:
I've been involved with getting better data for running Asterisk on
the Amazon EC2 cloud computing
James Moore wrote:
Notice that one of the prohibited items is:
# Phone Services - includes 800 or 900 phone services and audio text
services, prepaid phone cards, and prepaid phone services.
https://payments.amazon.com/sdui/sdui/about?acceptableuse
Google Checkout started with these
Ira wrote:
At 09:30 AM 1/27/2009, you wrote:
People are always going to ask stupid questions.
For me it's not so much the stupid questions as the expectations that
we're here to solve their problems according to their needs. If that
continues to happen and the noise level gets
Steve Edwards wrote:
On Thu, 22 Jan 2009, Wilton Helm wrote:
If some of your directories like /home and /user have separate mount
points, they don't have to get wiped out in the process.
If there is any reason to suspect a hack, re-installation is the only way.
I would replace
Take a look (if it still exists) at the Asterisk B2BUA project. It has a
patch that adds direct access to SIP response codes. It takes a little
modification of the patch file to use in some of the newer asterisks
(and to strip out the one codec option that's somewhat irrelevant), but
it's
and more carriers are offering SIP trunking
to their wholesale customers, which means that VoIP providers themselves
can now pick up the traffic over the Internet or via a dedicated private
IP link without having to deal with all that TDM stuff. This lowers the
barriers to entry and capital
Alex Balashov wrote:
SIP wrote:
What's interesting is the number of caveats and mixes even in the CLEC
and ILEC world. I work with a CLEC that is also an ILEC (in certain
areas), since they encompass various areas in Georgia (and own the
state's largest contiguous network, passing
Look, ma... spam! We dun never seen that 'n before.
N.
Sunkara RaviPrakash wrote:
Hi,
Imagine a billion Indians together.
Already 3 million Indians have chosen Indyarocks.com to bring India
together.
I am already part of it and dont be surprised if you find most of your
other
*From:* asterisk-users-boun...@lists.digium.com on behalf of Philipp
Kempgen
*Sent:* Thu 18/12/2008 4:17 PM
*To:* Asterisk Users
*Subject:* Re: [asterisk-users] Dial timeout with SIP - how to set
timeout for INVITE
It's a valid concern, but be prepared for people to tell you that this
should be done with the qualify parameter to determine if a host is up
and running. Not the most ideal way to handle it, I'll agree. But the
SIP proxy functionality of Asterisk is limited (as it's not intended to
be a SIP proxy
Steve Edwards wrote:
On Wed, 17 Dec 2008, Danny Nicholas wrote:
OSUR GONNA BE ABLE TO MAKE PEOPLE STOP POSTING. IF DIGIUM GETS ENOUGH OF
THESE STUPID HITS, THEY WILL CUT THIS OFF. I KNOW I'M SHOUTING, I'M
@#$###$# TIRED OF INTERRUPTING IMPORTANT WORK TO READ NOTHING. THAT'S WHAT
MSN
Jeff LaCoursiere wrote:
On Sun, 14 Dec 2008, Tzafrir Cohen wrote:
Right. So for those of us who want to do simple things and avoid
complicated stuff such as telephony in shoddy continent of North
America, could you please provide data for your country?
So far we have AU, IL and NZ.
Michael wrote:
Yes, but with an A-Z carrier, this can become risky when landline calls are
charged very differently to cellular calls, as is the case in NZ, Australia
and many other countries, unless someone is just a 'virtual' provider and
letting their up line do the invoices.
Some
Fred Posner wrote:
On Dec 5, 2008, at 11:31 AM, Michael Graves wrote:
On Fri, 05 Dec 2008 10:59:54 -0500, Neil Fusillo wrote:
Michael,
Was there something particularly special you had to do to get your M3 to
work? I'm now on my second one from E4 Technologies (from whom I'm still
waiting
at
http://VoipUsersConference.org including info on a SipAddHeader()
kludge to avoid DTMF problems.
IRC is Freenode.net #voip-users-conference join this even if you
can't call in.
Call via SIP: [EMAIL PROTECTED] (thanks to OnSip.com)
Call via PSTN (724) 444-7444 DTMF 22622# 1#
or try
Doug wrote:
At 18:56 12/1/2008, Tilghman Lesher wrote:
On Monday 01 December 2008 06:21:33 pm Doug wrote:
We tell our customers that they are not allowed to
download copyrighted material.
So your customers are only allowed to download public domain
material? That kind of restricts
Doug wrote:
At 04:03 12/2/2008, Benny Amorsen wrote:
Doug [EMAIL PROTECTED] writes:
Net Neutrality is great in principle. But ISP's need to
somehow control those few percentage of users who suck down
a huge majority of the bandwidth. It's dollars and cents.
Yes, just like the
Doug wrote:
At 07:00 12/2/2008, SIP wrote:
Doug wrote:
At 18:56 12/1/2008, Tilghman Lesher wrote:
On Monday 01 December 2008 06:21:33 pm Doug wrote:
We tell our customers that they are not allowed to
download copyrighted material.
So your customers are only allowed
Tom Browning wrote:
FWD (Free World Dialup) allows any SIP call to US toll free numbers
via [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
This works WITHOUT the need to be registered at FWD so in my dialplan
I have something like:
exten = _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r
http
Unless the LB is SIP-aware, and can maintain a SIP session, I don't see
how it would work. As the SIP command stream sends discrete commands,
without some sort of basic level of session awareness, there's no
guarantee over a reasonable-length call that the INVITE and BYE would
even get sent
all running a slightly
modified BSDI. And only slightly modified in packaging.
As for the current F5 SIP load balancer, we tried it a few years back
and it was a dismal failure. It wanted to do cookie-based SIP load
balancing and only worked with certain SIP proxies.
N
SIP
(3 trunks now, instead of 2) and having all 3 in use is not an issue.
Problem: Make a call on a Polycom 320 IP phone to any number and (4/5
times) it will drop the call after 30 seconds. I noticed that the
little timer that pops up on the LCD on the phone is missing when a
call
Greg Woods wrote:
On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote:
Gotta love this list being farmed for spammers now. I am sure they call
it targeted delivery or some such nonsense. I can't wait for capitalism
to completely fail, then there won't be any spam.
Socialism
randulo wrote:
On Tue, Nov 4, 2008 at 5:00 PM, Michael Graves [EMAIL PROTECTED] wrote:
In any case, the wideband bridge for this weeks VUC call supports only
G.722.
But we do plan to make a recording of both conference version available,
AFAIK?
r
But will it be a high-def
Joseph wrote:
I'm using Linksys SPA3102 adapter and have a strange ring tone:
Long-Short-Short or Long-Long-Short-Short
Does anybody know which setting adjust this ring tone on SPA3102
Sipura rings normally. I'm not sure if setting are on Regional Tab or User Tab
Interestingly, I get
/OpenSIPS more
than SER.
Are there some hard reasion for this.
I am in the process of deciding which SIP server i want to use with
Asterisk and just made a go at SER. Compilation was a little rough but
it was manageable. I threw away every module which funtionality i didn't
wanted
Alex Balashov wrote:
SIP wrote:
Seriously, though... this seems to be a popular misconception. I hear it
a lot. Where did you come across the information that SER is no longer
developed?
That seems to be a consequence of looking at the releases.
Anyway, I spoke too soon
It's not 100% secure. Like any dual-key encryption, it's subject to the
classic man-in-the-middle attack. This is why the Windows Zfone app has
the addition of a visual key you can read and coordinate with the
recipient to determine if a MITM attack is occurring. But only if you
know what you're
it, but not being the world's most
proficient C coder, I'm always afraid I'll break something else. ;)
N.
Andrew Joakimsen wrote:
I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
public with no NAT... everything works on the Asterisk end just fine
EXCEPT that I can never check voice mail
Philipp Kempgen wrote:
Andrew Kohlsmith (lists) schrieb:
On October 5, 2008 12:22:37 pm Philipp Kempgen wrote:
---cut---
http://lists.digium.com/pipermail/asterisk-users/2008-October/219538.html
http://lists.digium.com/pipermail/asterisk-users/2008-October/219541.html
Eric ManxPower Wieling wrote:
Olivier wrote:
I don't have any spare zaptel enabled system I could try this on, but I
was not aware of this CHANNEL variable.
Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables
Maybe, I will add a line in www.voip-info.org
gateway, your machine is misconfigured and internet traffic will not
properly flow.
I know you're just the messenger here, and it's not your fault. But the
message is wrong. Ekiga has tried to solve a problem (that of
determining a 'best path' for SIP to allow data flow in a NAT or
filtered scenario
Brian J. Murrell wrote:
On Fri, 2008-09-26 at 10:16 -0400, SIP wrote:
The RFCs are there for a reason. All SIP forking is UAS territory. Not
UAC territory.
From http://bugzilla.gnome.org/show_bug.cgi?id=553810 Damien Sandras
asks:
I repeat, Ekiga is doing something
My thoughts are that to do parallel requests from every IP address on
the machine is extremely weird behaviour.
How would any server know which to respond to?
SIP forking is supposed to send requests to multiple different
destinations (or fork mid-stream to send to different destinations
of packets. Figure out what to do with them. I'll be waiting for
your response.
There's a reason routing rules exist and mature services allow you to
control the interface from which it originates.
N.
Brian J. Murrell wrote:
On Thu, 2008-09-25 at 14:56 -0400, SIP wrote:
Sending from
Alex Balashov wrote:
You need to define what you mean by SIP forking. There are many
things people mean by that. They are usually one of:
1) Call branching (proxies do this).
2) Parallel but distinct call legs managed by a UAC (this is what
Asterisk does when you Dial(SIP/exten1SIP
Olivier wrote:
Hi,
A somehow related question, is broadcasting streaming music as music
on hold, submitted to any licencing fee ?
Regards
___
-- Bandwidth and Colocation
It's common sense. Using all iLBC, I can't seem to get 100 simutaneous
calls on my AMD 486 dx2/66.
I don't get it! ;)
N.
Eric ManxPower Wieling wrote:
Where did you hear this?
Shaun Wingrin wrote:
I have heard it said that, Asterisk falls over at 100 simultaneous
calls. Is this
Hi Sean,
Worked like a charm, thanks so much for the help!
On Sat, Sep 13, 2008 at 6:48 AM, Sean Bright [EMAIL PROTECTED] wrote:
Sip Support wrote:
exten = s,1,Set(CALLERID(name)=${CALLERIDNAME} AppropriateTag)
Try:
exten = s,1,Set(CALLERID(name)=${CALLERID(name)} AppropriateTag
Really? I thought both IAX and SIP are, at 3 characters apiece, equally
short.
However, if you get into IAX2, then yes... SIP is definitely a shorter
answer.
N.
Alex Balashov wrote:
The short answer is SIP.
Stefan Gofferje wrote:
http://www.voip-info.org/wiki-IAX
http://www.voip
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