I think Brendan is asking about endpoints (how to connect fax machines to
pure VoIP).
Short answer:
- you could connect standalone T.38-enabled analog gateways to 1.4,
- with 1.6, you can also use an analog board inside a server and connect fax
machines to this board.
hello:
i want to test the g729 with asterisk. my scenario is sipp(ulaw)-asterisk1
with g729-asterisk2 with g729.
I want to test g729 module with asterisk-1.4.21, when i make calls from
asterisk 1 to asterisk 2, the asterisk 1 always send ulaw to asterisk 2. my sip
in asterisk 1 is with codec
Craig Van Ham wrote:
I had weird issues when using a Sonicwall, gave up.
Same here, avoid them! I use the SnapGear SG560 now.
- Mike
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On Thu, 23 Oct 2008, Nhadie wrote:
hi,
hs anyone able to make video to work on asterisk? i tried following this:
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam
i can see that eyebeam is trying to broadcast a video but the other
eyebeam is not receiving it.
Gordon Henderson wrote:
On Thu, 23 Oct 2008, Nhadie wrote:
hi,
hs anyone able to make video to work on asterisk? i tried following this:
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam
i can see that eyebeam is trying to broadcast a video but the other
eyebeam
Hi,
I've been very puzzled lately. I installed a phone system for a friend
a few weeks ago, and they're having a problem that I can't get rid of,
actually 2 problems. Before I go into the problems, let me tell you
about the setup. It's a pretty small setup with only 4 handsets, all
Polycom 320s,
For a door opener on an Astribank FXS port we need a loop current of 24.5mA .
It does not function with the Astribank now, the dialtone becomes quiet
immediately after pressing the button on that device.
I've seen a limit of 23mA in the zaptel source.
Is it possible to change the loop current of
On Thu, 23 Oct 2008, Nhadie wrote:
Gordon Henderson wrote:
On Thu, 23 Oct 2008, Nhadie wrote:
hi,
hs anyone able to make video to work on asterisk? i tried following this:
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam
i can see that eyebeam is trying to
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nhadie
Sent: Thursday, October 23, 2008 6:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk video
Gordon Henderson wrote:
On Thu, 23
Hi sir,
sorry typo on the e-mail, this is what i have:
sip.conf
[100666]
type=friend
secret=666
allow=h263
allow=all
dtmfmode=rfc2833
canreinvite=no
host=dynamic
context=testvideo
nat=yes
[100777]
type=friend
secret=777
allow=h263
allow=all
dtmfmode=rfc2833
canreinvite=no
host=dynamic
Hello everyone!
I've just switched from Asterisk 1.6.0-beta9 to 1.6.0.1 and my mISDN is not
working. Here's what happens, if I try to call the line:
bach P[ 1] -- !! lib: No free channel!
P[ 1] -- we have already send Release_complete
I haven't changed the configuration fles. Should I
Hi, which version of asterisk are you running? Perhaps if you post your
extensions.conf and others related files you could get more accurate help.
If you answer a ringing phone and you can't answer the call, there you
could have a network or sip config problem, that means that the SIP packet
Does the call-limit directive work on those SIP items defined in users.conf as
it relates to presence and queues?
Jeremy Mann
Director of IT
Texas Health Management Group
Direct Line: 817-310-4956
Main Line: 817-310-4999
Fax: 817-310-4990
Email: [EMAIL PROTECTED]
Hi,
We are testing new Asterisk 1.6.0.1 because we would like to use the
Attended Transfer feature and we are trying to use the new action Atxfer
developed for AMI.
As far as we know, it is suposed to be in this release as it can be read
in Digium's changelog
/New command: Atxfer. See
David Monteagudo Sanz wrote:
Hi,
We are testing new Asterisk 1.6.0.1 because we would like to use the
Attended Transfer feature and we are trying to use the new action Atxfer
developed for AMI.
As far as we know, it is suposed to be in this release as it can be read
in Digium's
Have an interesting problem,
Using asterisk 1.6.0.1
Phone A receives voicemail, dials into VoiceMailMain, Phone B's BLF for
A lights up.
Phone A deletes the voicemail but still in VoiceMailMain, Phone B's BLF
for A goes off.
Phone A hang's up, Phone B's BLF for A goes on.
From this point
Hello all,
I have an asterisk box running in a customer with Hylafax, iaxmodem,
asterisk 1.2.18.
The service can receive faxes, from a lot of fax machines, but there
are a couple of them that asterisk Hylafax cannot complete.
This calls arrive the asterisk box, asterisk detect that this calls
Suddenly my system crash whem I see core show channels are increasing until
reaches its limit at asterisk.conf
It seems channels (Local, Zap, SIP) are not being closed.
The problem persists and I don't know what to do
Please help me!
___
-- Bandwidth
And this phone are connected in a local LAN??
Because I see Asterisk receiving a Bad request from 68.156.63.118
If those phones are not in your local LAN, try with a soft phone first.
Could be Zoiper or Xlite.
Besides, use SIP SET DEBUG, for SIP debugging and try to see why is 101
sending a 400
Sorry for asking the obvious question, but are there other elements of
the slow path besides the Sonicwall? I mean, what is in front of the
Sonicwall? Also, might the Sonicwall be positioned as some kind of choke
point in the topology, thus leading to genuine sporadic congestion?
James
voip crazy wrote:
This calls arrive the asterisk box, asterisk detect that this calls
are fax, asterisk answer the call, and then Hangup the call. But
hylafax do not receive nothing.
What does the CLI say about this call?
Thanks,
Lee.
___
--
My version is 1.4.21.1
On Thu, Oct 23, 2008 at 11:38 AM, Daniel - Asterisk [EMAIL PROTECTED]wrote:
Suddenly my system crash whem I see core show channels are increasing until
reaches its limit at asterisk.conf
It seems channels (Local, Zap, SIP) are not being closed.
The problem persists
We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP
connections. I've seen the delay thing, as well as the Sonicwall throwing
away entries from the ARP table because of inactivity. I've also seen
sporadic, intermittent problems with transfer from one phone to another.
I
Lee Howard wrote:
voip crazy wrote:
This calls arrive the asterisk box, asterisk detect that this calls
are fax, asterisk answer the call, and then Hangup the call. But
hylafax do not receive nothing.
What does the CLI say about this call?
With high verbosity, I might add.
--
Alex
Bill Michaelson wrote:
Sorry for asking the obvious question, but are there other elements of
the slow path besides the Sonicwall? I mean, what is in front of the
Sonicwall? Also, might the Sonicwall be positioned as some kind of choke
point in the topology, thus leading to genuine sporadic
I'm restarting my system without solution and I've extended my call limit to
10 calls (asterisk.conf) to avoid call rectriction.
But, why now? It was working well from July until this morning.
Thanks in advance for every help you can give.
Daniel
On Thu, Oct 23, 2008 at 12:04 PM, Daniel -
Can you get another public IP? If so put another router in. Use vlans
to seperate the traffic.
Sent from my iPhone
On 23-Oct-08, at 11:28 AM, James Lamanna [EMAIL PROTECTED] wrote:
Bill Michaelson wrote:
Sorry for asking the obvious question, but are there other elements
of
the slow
On Thu, Oct 23, 2008 at 12:39 PM, Juan RodrÃguez [EMAIL PROTECTED] wrote:
And this phone are connected in a local LAN??
Because I see Asterisk receiving a Bad request from 68.156.63.118
If those phones are not in your local LAN, try with a soft phone first.
Could be Zoiper or Xlite.
Besides,
On 10/23/08, Bruce Komito [EMAIL PROTECTED] wrote:
We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP
connections. I've seen the delay thing, as well as the Sonicwall throwing
away entries from the ARP table because of inactivity. I've also seen
sporadic,
Indeed I am going for pure voip and trying to figure out how to
implement t.38, as you suggest.
On Oct 23, 2008, at 2:08 AM, Olivier wrote:
I think Brendan is asking about endpoints (how to connect fax
machines to pure VoIP).
Short answer:
- you could connect standalone T.38-enabled
You're absolutely right. I only mention Sonicwall, because those are the
ones we see most often and there is a perception out there that, because
Sonicwall is the (disputed) leading firewall, it should work.
Bruce Komito
WPTI Telecom
(775) 236-5815
On Thu, 23 Oct 2008, Kristian Kielhofner
On Thu, 2008-10-23 at 09:36 -0700, Marc Hudson wrote:
I've looked at 'core show hints' and it is in fact reporting INUSE when
it's not, and NOT_INUSE
when it is.
That definitely sounds like a bug to me. Could you please report this
on the bug tracker, so that the developers can take a look
Jared Smith schrieb:
On Thu, 2008-10-23 at 09:36 -0700, Marc Hudson wrote:
I've looked at 'core show hints' and it is in fact reporting INUSE when
it's not, and NOT_INUSE
when it is.
That definitely sounds like a bug to me. Could you please report this
on the bug tracker, so that the
Maybe I just haven't thought of the right google search terms -- but is
there a website/guide out there that will help me understand the output
from pri debug span?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
You would be much better off trying this question on the asterisk-users
list.
Much more traffic and geared towards Asterisk in general :)
-Brandon
John Cheng wrote:
Maybe I just haven't thought of the right google search terms -- but is
there a website/guide out there that will help me
Hello all,
I've got dialout= and callback= set in my voicemail.conf so that I
can have users return calls to folks who have left messages. They
really like this feature.
But when the callback is over, a normal hangup occurs instead of the
caller being put back into voicemail at the next message.
On Oct 23, 2008, at 3:10 PM, John Cheng wrote:
Maybe I just haven't thought of the right google search terms -- but
is
there a website/guide out there that will help me understand the
output
from pri debug span?
___
perhaps this might be helpful?
Q.931
Hey folks,
I am involved with a group that is going to use Twitter, SMS, iPhone,
and Asterisk to field-monitor the upcoming US elections.
The group is pretty large scale and you can find out more here:
http://votereport.pbwiki.com
We need some help with SIP telephony infrastructure.
Hello Dave,
We can offer you. What area DID you are looking for.
Jai
Buy SIP DID, www.didforsale.com
On Thu, Oct 23, 2008 at 2:20 PM, David Troy [EMAIL PROTECTED] wrote:
Hey folks,
I am involved with a group that is going to use Twitter, SMS, iPhone, and
Asterisk to field-monitor the
I am able to now call the second extension when setup like this so I
believe I'll leave it alone for a while. Basically added the extension
102 to the main incoming line:
exten = 101,1,Dial(SIP/101SIP/102SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?lbl_default_1:)
Dear Sir,
Please let us know which specific location you need and we can offer that
service for you
Regards
On Fri, Oct 24, 2008 at 12:20 AM, David Troy [EMAIL PROTECTED] wrote:
Hey folks,
I am involved with a group that is going to use Twitter, SMS, iPhone, and
Asterisk to field-monitor
We have a number of DID's that do the standard VoIP tricks: ringing
multiple locations, findme-followme etc. What is happening more and
more is that customers call those DID numbers, and draw the reasonable
conclusion that they are calling mobile numbers because they literally
can HEAR that the
hello,
I am using sip, my default codec is set to gsm in sip.conf
Using call files, is there a way to send out a call using
ulaw while other channels are using gsm ?
tia.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
I was thinking about complicating my Voip setup by adding CME. I found
this example here:
http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration
and here: http://www.pasewaldt.com/cme/cme_index.htm
Would anyone like to comment on their experiences using CME with
Hi, Sorry I forgot to mention versions and post files.
Asterisk version:
pbx:/etc/asterisk# asterisk -rx core show version
Asterisk 1.4.22 built by root @ coope-pbx on a i686 running Linux on
2008-10-22 09:36:35 UTC
I'm running zaptel 1.4.12.1 and wanpipe 3.3.14. Also tried zaptel
1.4.11 and
Dare I ask why you want to do this?
Dave
On Oct 23, 2008, at 10:00 PM, Stephen Reese wrote:
I was thinking about complicating my Voip setup by adding CME. I found
this example here:
http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration
and here:
On Thu, Oct 23, 2008 at 10:57 PM, David Gibbons [EMAIL PROTECTED] wrote:
Dare I ask why you want to do this?
Dave
I know it seems counter intuitive but I've several examples of it
being done and for me it would be for the experience of working with
CME. A lot of companies utilize Cisco
On Wed, 2008-10-22 at 14:56 -0500, Terry Wilson wrote:
hi
for any context ,you must to open /etc/asterisk/extensions.conf and
insert this line : exten =Realtime/[EMAIL PROTECTED]
and (reload) or (restart now) your asterisk
You don't have to restart asterisk, just a 'dialplan reload'
Philipp Kempgen wrote:
Jared Smith schrieb:
On Thu, 2008-10-23 at 09:36 -0700, Marc Hudson wrote:
I've looked at 'core show hints' and it is in fact reporting INUSE when
it's not, and NOT_INUSE
when it is.
That definitely sounds like a bug to me. Could you please report
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