Mike,
This firmaware works on Buffalo, linksys and some asus routers.
Linksys did release the wrt54gL because of the demand to have a router
with Linux.
In fact, the L means Linux and this router is still in production, easy
to find (in Europe anyway) and very very cheap.
DD-Wrt also runs
I had a patch created for 1.4.X for this.
http://bugs.digium.com/bug_view_page.php?bug_id=14159
- Original Message -
From: Gabriel Ortiz Lour
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 18, 2009 8:23 PM
Subject: [asterisk-users] Global h exten
Hi all,
I had a patch created for 1.4.X for this.
http://bugs.digium.com/bug_view_page.php?bug_id=14159
- Original Message -
From: Gabriel Ortiz Lour
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 18, 2009 8:23 PM
Subject: [asterisk-users] Global h exten
Hi all,
Hi,
I was asked for the patch and I sent it. Does anybody have any news about
this subject?
I'm willing to try a fix for 1.4 but I'd need any guidelines to do it.
Thanks in advanced
Jose
2009/4/2 Moises Silva moises.si...@gmail.com
Async AGI was never released for Asterisk 1.4.X, so probably the
Hi,
I know it is a bit off-topic, but I'd like to ask the community what is the
current most supported way to deal with DTMF?
I'm looking for an all-SIP system and I'm mostly interested in the end
devices support of the different methods (DTMF in-band audio, DTMF RTP
telephony events packets, SIP
Hi,
we are using version 2.0.4 (vicidialnow distribution) now for some time
in productino - working quit nice.
Is there any upgrade instruction out there - or will a simple yum update
do the job in the feature.
PS: On the astguiclient site you have April 3, 2008 -
Released version 2.0.5 - i
Dear Martin
Can you inform me how to make the patch or from where I can get it otherwise
if there is an application can generate it?
Or if its relate to chan_sip.c ,please can you tell me which function to
edit or lines to be added
Regards
-Original Message-
From:
On Mon, 6 Apr 2009, Tzafrir Cohen wrote:
On Sun, Apr 05, 2009 at 11:35:18PM +0200, Puskás Zsolt wrote:
On Sunday 05 April 2009 21.28.48 Gergo Csibra wrote:
Saturday, April 4, 2009, 3:13:12 PM, Puskás wrote:
Got it working with Asterisk 1.2 installed on the same PC as Asterisk 1.4
[ in
Hi Philipp!
On Sunday 05 April 2009, Philipp von Klitzing wrote:
Take a look at these two links:
Thanks for the links! So one option is to implement domain based
authentication, which would be quite a bit of work. Another option
which is quite popular is using an openSER (one of the two forks)
Implementing support for configuration of skills using an XML file would
require rewriting one function. Adding the skill selections as an
option of the queue would require a few lines of code. Apart from that
your proposal pretty much matches my implementation.
Cheers,
Florian
On
Thanks for the tip, Harry. I will try that when I have exhausted all
avenue. My problem is that if I upgrade to 1.4.24 and DAHDI, I'll break
other stuffs.
In my current set up, the PRI did work for a long period of time (7
hours) before going into this unreliable mode (up and down). I'm
Mark Michelson wrote:
Caution: One shortcoming of queue member penalties is that they are not
taken into account if a queue member of a low penalty does not answer a
call. Say for instance that the queue application determines that there
are two members available to answer an incoming call.
HI,
Recently, I found that asterisk fail to get the correct context of
the sip phone. Below is the configuration and the log message. In
the log message, asterisk fail to identify the calling party. As a
result, it use a default context instead of int. Anyone know why and
how to fix it?
On 4/6/09, Wolfgang Pichler wpich...@yosd.at wrote:
Hi,
we are using version 2.0.4 (vicidialnow distribution) now for some time
in productino - working quit nice.
Is there any upgrade instruction out there - or will a simple yum update
do the job in the feature.
PS: On the
Hi all,
Lastly we are getting several of the following errors:
app_queue.c: No one is answering queue
And when you isse a queue show XXX the status of the peers are reported as
Invalid.
We tried 1.4.23.1 and reverted back to 1.4.18.1 because it has showed good
behaviour in the past but no
Hi all,
I use a mysql table for sip users and I fixed rtcachefriends param to yes
in order to have a caching of this table.
I would like to know how often does Asterisk check the mysql table to update
its caching please.
Regards,
Cédric.
--
Cédric Bonnet
/FT/NCPI/DPS/CTR/CPM/VASF
Tel.
If that someone is between you and the other endpoint (like between you
and the switch, or using port-mirroring on a router somewhere), then
yes. The conversations can be recorded. In the US, the ability to be
able to do this is required by law. You've little to worry about random
hackers coming
Martin escreveu:
What is the specification for T309 ? I'm too lazy to look it up.
The default behaviour when the alarm of layer 1 (electrical T1/E1) is
detected is to assume
all calls dropped on both sides and that's what Asterisk does.
The timer is simply deactivated since all the calls
Any hardware that can do 25-50-100 fxs ports trunked to sip ?
Example one end a cat5 other end 50 RJ11's jacks..
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To UNSUBSCRIBE or
I have an ITSP we are trying to work with that has an Unusual way of
working, but that said my understanding of their behaviour is that it
is fully RFC compliant. Can someone suggest how I might be able to
interoperate under these circumstances:
We register fine with them, and send the default
Hello,
I want to set up a Voip Farm (c) (tm) (patent pending) but don't know
how to do it.
Please help.
Oh, the irony :)
Cheers
Jean-Michel.
2009/4/2 Gabriel - IP Guys gabr...@impactteachers.com:
Dear All,
Thanks for taking the time to read this. I have been presented with a massive
Does IPKALL still exist?
I am after a free SIP trunk - who is still giving these away these days?
As I noticed Stanaphone is no longer in business?
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net +1-212-203-4357 New York
+61-2-9016-5642 (Sydney
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
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... :(
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.
N.
Dean Collins wrote:
Does IPKALL still exist?
I am after a free SIP trunk – who is still giving these away
None of their pages apart from the front page seem to work though
http://phone.ipkall.com/ipphone/login.asp
Are you sure they still exist?
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
+44-20-3129-6001 (London
Hello,
For corporate users, how would you define Call Forwarding services ?
1. Would offer option A or B ?
option A:
no forwarding
immediate
busy
no answer
option B:
no forwarding
immediate
busy
no answer
busy or no answer
I've seen legacy PBX offering B and SIP phones offering A.
Which is the
On Mon, 6 Apr 2009, ContactTel Business wrote:
Any hardware that can do 25-50-100 fxs ports trunked to sip ?
Example one end a cat5 other end 50 RJ11's jacks..
Audiocodes MP-124 has 24 FXS ports and an amphenol connector (so you would
need to add a breakout box to get your RJ11 jacks).
j
On 6 Apr 2009, at 14:32, Dean Collins wrote:
None of their pages apart from the front page seem to work though
http://phone.ipkall.com/ipphone/login.asp
http://phone.ipkall.com/login.asp
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Hmm, this seem to be the biggest non cisco device i found as well,
The breakout is a FXS, 50-pin Telco to rj11 converter ?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: April-06-09
On Mon, 6 Apr 2009, ContactTel Business wrote:
Hmm, this seem to be the biggest non cisco device i found as well,
The breakout is a FXS, 50-pin Telco to rj11 converter ?
Yes. MP-124 is a solid, stable device. Any vendor that sells you the
MP-124 will have a breakout box (or patch panel)
Grandstream GXW4024 IP Analog Gateway
Also seem to do it, not sure what is better between AC and GS..
I think AC is more complicated to program but better quality, while GS is
half the price of the AC.
But also comes with rj11 jacks.. the AC has a 50 pin
Any opinions ?
Basically need to wire
I registered few days back and got a DID. Maybe this is temporary ?
On Mon, Apr 6, 2009 at 7:05 PM, Steve Howes st...@geekinter.net wrote:
On 6 Apr 2009, at 14:32, Dean Collins wrote:
None of their pages apart from the front page seem to work though
On Mon, 6 Apr 2009, ContactTel Business wrote:
Grandstream GXW4024 IP Analog Gateway
Also seem to do it, not sure what is better between AC and GS..
I think AC is more complicated to program but better quality, while GS is
half the price of the AC.
But also comes with rj11 jacks.. the AC
Actually thinking about it, that 50 pins is simply the 48 + 2 grounds i
imagine.. or something of the likes..
Thanks Jeff, you have pointed me in the right direction.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
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...
I have a few questions.
Asterisk is a windows program why each time I try to find out how
communicate with my Panasonic TDA 100 or with TDE 100 always read use
one card o use a box why I can't use simply my network card, in the
other side of Panasonic exist two types of cards one in TDA 100
jibanez1...@cimex.com.cu wrote:
I have a few questions.
Asterisk is a windows program
Asterisk is not a windows program.
why each time I try to find out how communicate with my Panasonic TDA
100 or with TDE 100 always read “use one card o use a box” why I can’t
use simply my network
Take a look on xorcom solutions
http://www.youtube.com/watch?v=qt4aPdGIvIQfeature=player_embedded
Regards,
Luis Morales
On Mon, Apr 6, 2009 at 8:09 AM, ContactTel Business
li...@contacttel.com wrote:
Any hardware that can do 25-50-100 fxs ports trunked to sip ?
Example one end a cat5
You have to understand that this mailing list is not free instant
support. Even more, you are using an unsupported Asterisk feature for
1.4. I will check it when I have some spare time to try to reproduce
and fix it. If you are too much in a hurry you can always contact me
off-list for paid
Dears
Asterisk is a median server between the caller and the terminations gateway
The caller send the call to asterisk à asterisk will play music on hold
untill the termination gateway send 200 OK and the RTP establish
My problem that, Asterisk is not forwarding the 180 ringing from the
On 6 Apr 2009, at 15:40, Khaled W. Chehab wrote:
Dears
Asterisk is a median server between the caller and the terminations
gateway
The caller send the call to asterisk à asterisk will play music on
hold untill the termination gateway send 200 OK and the RTP establish
My problem
On Sat, Apr 4, 2009 at 11:18 AM, Timothy Smith timotsm...@gmail.com wrote:
We're migrating from Cisco to asterisk because cisco is expensive to
maintain, besides we can achieve more with asterisk like customised
IVRs etc.
I don't know what expensive to maintain means. We spend more on our
cyr2...@gmail.com schrieb:
This message was sent on behalf of cyr2...@gmail.com at openSubscriber.com
Use the appropriate header field for that information.
It's called From (in contrast to Sender).
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Based on the Asterisk logs you posted the Asterisk doesn't have it
implemented per:
The implementation of timer T309 in the user side is optional
Martin
On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann afo...@disc-os.org wrote:
Martin escreveu:
What is the specification for T309 ? I'm too
Hi,
The easiest is to turn off MOH on the Dial. Otherwise the patch is
easy but not trivial.
Once the B-leg receives the ringing message and passes it in Dial app
then the code has to turn off the MOH
and tell the A-leg to send the ringing message. At the same time the
code that skips passing the
Are there an IOS interface for Asterisk?, or an IOS to SIP converter?
Some femtocells uses this protocol and I would to use them with Asterisk.
Jorge Mendoza
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asterisk-users
It's SIP in rfc (RFC2833) then SIP INFO and then if you can't do
anything else inband audio (only G711)
Martin
On Mon, Apr 6, 2009 at 2:24 AM, Cesc Santa cesc.sa...@gmail.com wrote:
Hi,
I know it is a bit off-topic, but I'd like to ask the community what is the
current most supported way to
Have you looked at the syntax of register = keyword ?
register = [transport://]user[:secret[:authuse...@host[:port][/extension]
; If no extension is given, the 's' extension is used.
There you have it ... Contact: sip:s
set the extension and you should be fine
Martin
On Mon, Apr 6, 2009
That's because you have to create a user account in sip.conf ... +
Asterisk is sensitive about it.
What should help is if you register the phone with that sip account first.
Martin
On Mon, Apr 6, 2009 at 5:00 AM, Rilawich Ango maillist...@gmail.com wrote:
HI,
Recently, I found that asterisk
Thanks for the reply - Perhaps I was not clear.
On the register= line, if I set /extension to be /12345, then this
just replaces 's' with 12345, and ALL calls, regardless of their
destination number will be routed on the INVITE line to 12...@x.x.x.x,
and the actual destination is specified in the
Why would i want to do that ?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
Sent: April-06-09 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Jorge Mendoza wrote:
Are there an IOS interface for Asterisk?, or an IOS to SIP converter?
Some femtocells uses this protocol and I would to use them with Asterisk.
Jorge Mendoza
___
You're comparing to apples to Orange. IOS is the Cisco
Martin escreveu:
Based on the Asterisk logs you posted the Asterisk doesn't have it
implemented per:
"The implementation of timer T309 in the user side is optional"
Martin
On Mon, Apr 6, 2009 at 6:22 AM, Afonso Zimmermann afo...@disc-os.org wrote:
Martin escreveu:
What is the
This may be your solution.
Regards,
Luis Morales
On Mon, Apr 6, 2009 at 12:23 PM, ContactTel Business
li...@contacttel.com wrote:
Why would i want to do that ?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Aint this based on asterisk ?
I don't think i would use that, thanks anyway.
And yes i know this is an asterisk list.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Morales
Sent: April-06-09 2:02 PM
Jorge Mendoza wrote:
Brent Davidson wrote:
Jorge Mendoza wrote:
Are there an IOS interface for Asterisk?, or an IOS to SIP converter?
Some femtocells uses this protocol and I would to use them with Asterisk.
Jorge Mendoza
___
Actually i might rephrase, i need hardware solution not pc based, no hard
drives, no fans, no application you need to monitor, hence hardware,
You can ignore the rest of this thread i have my info.
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
David Ruggles schrieb:
I've done some googling and searched voip-info but I'm not able to find a
good answer about how to provision the GXP 2000.
Based on questions I've asked before it seems like a lot of people are using
the grandstream phones so I figure provisioning can't be that hard.
Brent Davidson wrote:
Jorge Mendoza wrote:
Are there an IOS interface for Asterisk?, or an IOS to SIP converter?
Some femtocells uses this protocol and I would to use them with Asterisk.
Jorge Mendoza
___
You're comparing to apples
Hi -
I just deployed a system using IMAP Voicemail. During my testing,
voicemail worked fine. I could check vm from the phone, and the
messages would get marked as read, or I could read the messages in a
mail client, and the phone's mwi light would turn off. Very neat.
I'm not exactly sure
Mr. ContactTel,
if you need hadware only take a look on:
http://www.telephonydepot.com/Catalog/Digium-TDM2400P/Digium-TDM2400P-Blank-Board
http://www.telephonydepot.com/Catalog/Digium-Accessories/Digium-S400M-Quad-FXS-Module
Hi, got a Sangoma A200 with a bunch of extension cards and having real
problems getting it to deal with a normal single BT line
Symptoms are that incoming calls are fine. Outgoing calls ring the far
end, BUT asterisk never sees that the call is answered (ie no message in
the logs files saying
The Asterisk Development Team is pleased to announce the fourth release
candidate of Asterisk 1.6.1.0. Asterisk 1.6.1.0-rc4 is available for
immediate download at http://downloads.digium.com/pub/asterisk/
This release candidate improves the performance of the ast_event cache
functionality, fixes
The Asterisk Development Team has announced the release of Asterisk 1.6.0.9.
Asterisk 1.6.0.9 is available for immediate download at
http://downloads.digium.com/pub/asterisk/
This release resolves a merge issue from trunk to 1.6.0.7 that caused memory
to be freed that should not be. In trunk,
Noah Miller wrote:
I just deployed a system using IMAP Voicemail. During my testing,
voicemail worked fine. I could check vm from the phone, and the
messages would get marked as read, or I could read the messages in a
mail client, and the phone's mwi light would turn off. Very neat.
I'm
Just FYI:
IP address 89.248.168.176 has been trying to use the recently release SIP
vulnerability in Asterisk to make outbound calls via our box. They are running
a bank account callback scam.
Jeremy Mann
Director of IT
Texas Health Management Group
Direct Line: 817-310-4956
Main Line:
Hi,
You're right. I wasn't aware of this patch getting into the code.
In the version you're running the code is already present.
The only problem I see is that some other timer kicks in here and the
T309 cannot be scheduled.
q931.c has this ...
/* For a call in Active state, activate T309 only
http://www.websiteoutlook.com/www.songania.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy Mann
Sent: April-06-09 3:55 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Hacked
Ok, I'll bite. What does websiteoutlook have to do with it?
The IP mentioned is in the Netherlands:
% Information related to '89.248.168.0 - 89.248.168.255'
inetnum:89.248.168.0 - 89.248.168.255
netname:NL-ECATEL
descr: AS29073, Ecatel LTD
country:NL
admin-c:
ping www.songania.com
PING www.songania.com (89.248.168.176) 56(84) bytes of data.
64 bytes from 89.248.168.176: icmp_seq=1 ttl=49 time=131 ms
If you clicked on it you would of seen it shows info on the domain, that is
hosted on it.. ill bite back ;)
Then on bottom.. Owned By Al-Sharif
Is there different points in the zaptel configuration according to each country?
Thanks.
_
Drag n’ drop—Get easy photo sharing with Windows Live™ Photos.
I'd read this article
(http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf) but as I see it,
you only have 2 lines in zaptel.conf for country specification; the rest of
the lifting is done in Zapata.conf.
_
From: asterisk-users-boun...@lists.digium.com
I'm not exactly sure when things got munged up, but something broke.
I can record messages with Voicemail(), but now when I access an IMAP
mailbox using VoicemailMain(), it always says there are no messages,
even when there clearly are (unread) messages in the IMAP mailbox.
This appears to
so they are only.
loazone
and
defaultzone
thanks.
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Mon, 6 Apr 2009 16:01:25 -0500
Subject: Re: [asterisk-users] Zaptel Config
I’d read this article
I have a server with 2 Lan Cards.
Now, when I am trying to make calls using Local Lan, its One way Audio which
means customer cant hear me but if I use Static IP with Wan Connection, it
works perfectly.
I changed the network from loc1 to loc2 but its same.
I tried changing Ethernet Card but no
Can it be that any Port got blocked ?
On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote:
I have a server with 2 Lan Cards.
Now, when I am trying to make calls using Local Lan, its One way Audio
which means customer cant hear me but if I use Static IP with Wan
Connection,
How tcpdump on interface show??
2009/4/6 David @ULC ucoms2...@gmail.com:
Can it be that any Port got blocked ?
On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote:
I have a server with 2 Lan Cards.
Now, when I am trying to make calls using Local Lan, its One way Audio
Few Running figures !!
On Tue, Apr 7, 2009 at 3:41 AM, David @ULC ucoms2...@gmail.com wrote:
I have a server with 2 Lan Cards.
Now, when I am trying to make calls using Local Lan, its One way Audio
which means customer cant hear me but if I use Static IP with Wan
Connection, it works
Can you give more information about this vulnerability ?
Martin
On Mon, Apr 6, 2009 at 2:55 PM, Jeremy Mann jm...@txhmg.com wrote:
Just FYI:
IP address 89.248.168.176 has been trying to use the recently release SIP
vulnerability in Asterisk to make outbound calls via our box. They are
First posted at:
http://deancollinsblog.blogspot.com/2009/04/australian-nbn-network.html
You ripper,
http://www.crn.com.au/News/100466,government-announces-nbn-plans.aspx
This is exactly how fiber should be - a shared wholesale resource just
like water to ensure retail channels
Thanks. Let me try it.
On Tue, Apr 7, 2009 at 12:23 AM, Martin asteriskl...@callthem.info wrote:
That's because you have to create a user account in sip.conf ... +
Asterisk is sensitive about it.
What should help is if you register the phone with that sip account first.
Martin
On Mon, Apr
Yeah some devices use callerid as user which is xxx in x...@deviceip
So if you see : chan_sip.c: Call from '1231231234' to extension
'5544' rejected because extension not found.
Then adding in the [user] stanza
user=foobar
fromuser=foobar
insecure=very ( or port,invite if that still alive)
On Mon, 6 Apr 2009, Dean Collins wrote:
http://www.crn.com.au/News/100466,government-announces-nbn-plans.aspx
This is exactly how fiber should be - a shared wholesale resource just
like water to ensure retail channels can deliver value add in content
and services;
- Not 'outspending on
Hellow
Can any body helps how can interfacing between asterisk and patton media
getway.
Thanks
mahboob
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Interface in what manner?
mahboob zaman wrote:
Hellow
Can any body helps how can interfacing between asterisk and patton media
getway.
Thanks
mahboob
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