Hi ! My name is Gerald and I am working with WEBRTC and JsSIP.I configure my Asterisk 11.7.0 to work wit WEBRTC.Using a JsSIP (http://tryit.jssip.net/), the SIP extension can connect at the Asterisk, but when we try to make a call they send a 488 response and finish it.here is the part of the SIP
Hi,
I am new in voip, how many calls can one asterisk box handle with 30 %
of trans-coded calls and system configuration as
8GB RAM
X3430 Xeon Processor, 2.4GHz, 8M Cache, Turbo
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Hello all,
I just instaled a tdm2400 Digium card on my asterisk box. When it
boots, I can see some error messages in dmesg.
wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 8 ms
in order to compensate.
wctdm24xxp :21:08.0: Missed interrupt. Increasing latency to 10 ms
in
Hello,
We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
This is the (relevant) test dialplan:
[incoming-private]
exten = _X., n, Dial(SIP/1001,30)
exten = _X., n,
Hello,
I want to analyze the asterisk logs files, looking for all kind of
errors, ¿Anyboby knows any asterisk logs analyzer?
Thanks all,
Voipcrazy
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Hello list members,
We're trying to get MWI notifications on our ATA device and we set it to
send SUBSCRIBE messages to Asterisk, but it gets UNAUTHORIZED messages,
despite the fact that we set the following lines in its settings in
sip.conf:
subscribemwi=yes
mailbox...@from-extensions
We
SendFax rejects T.38 reINVITE (488 Not
acceptable here)
From: Kevin P. Fleming kpflem...@digium.com
To: asterisk-users@lists.digium.com
Date: Thursday, 21 October, 2010 16:11:00
On 10/20/2010 11:35 AM, VoIP Question wrote:
On Wed, Oct 20, 2010 at 4:25 PM, Kevin P. Flemingkpflem...@digium.com
Not
acceptable here)
From: Kevin P. Fleming kpflem...@digium.com
To: asterisk-users@lists.digium.com
Date: Thursday, 21 October, 2010 16:13:02
On 10/20/2010 09:35 AM, VoIP Question wrote:
Thank you Kevin,
We'll upgrade our server to 1.6.2.12 and try again.
Another question: Is there (expect
Hello all,
We would like to inform the caller of the reason for a failed call.
For example, when we get a 486 Busy Here, the system accepts it and in the
CLI we see Everyone is busy/congested at this time.
Can we use this data to play an announcement to the caller?
Thank you in advance for
Thank you Kevin,
We'll upgrade our server to 1.6.2.12 and try again.
Another question: Is there (expect for the admin guide that we didn't
succeed to understand the example in) an example somewhere for ReceiveFax
full extensions.conf diaplan? We would like to allocate one of the
extensions that
Hello all,
We're trying to build a small IVR application to allow callers to use the
Asterisk for outgoing calls in a 2 steps dialing mode.
The context for outgoing calls is called outgoing (we have there an LCR
and routing mechanism we want to use, depending on the destination).
This is what
Hello again,
If I set a peer to use G.711 only, they try to process a sent fax in G.711,
but Asterisk doesn't like it:
WARNING[4903]: res_fax.c:1709 sendfax_t38_init: Audio FAX not allowed on
channel 'SIP/Main-000a' and T.38 negotiation failed; aborting.
What can I do to enable it?
Thanks,
Hello,
I'm trying to send a tif file, using Fax for Asterisk and the call is
executed, but when I get the reINVITE with T.38 data, the local server
doesn't recognize that we have this capability and sends a 488 message.
These are the logs:
--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---
It's set to yes for this peer.
also t38pt_udptl is set to yes.
:(
On Tue, Oct 19, 2010 at 5:12 PM, David Backeberg dbackeb...@gmail.comwrote:
On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question voip.quest...@gmail.com
wrote:
Hello,
I'm trying to send a tif file, using Fax for Asterisk
, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
Thanks.
Michael
On Tue, Oct 19, 2010 at 5:40 PM, David Backeberg dbackeb...@gmail.comwrote:
On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question voip.quest...@gmail.com
wrote:
It's set to yes for this peer.
also
Digium claims that their FFA is the best and most compatible solution and
they give one channel for free, but do not provide support for those that do
not buy more channels, but why buy more channels if the free/test one
doesn't work?
I know they read (and sometimes respond) to this list, so I
Max-Forwards: 70
Content-Length: 0
Thanks,
Michael
On Tue, Oct 19, 2010 at 8:56 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 10/19/2010 12:01 PM, VoIP Question wrote:
Digium claims that their FFA is the best and most compatible solution
and they give one channel for free, but do
Hello,
I would like to verify if a specific SIP header exists, and if yes, extract
the partial content from another header.
1. Is there a way to verify if a specific header exists?
2. How do I extract data that is between the first : and the following @?
Specifically, The data looks like
Hello all,
Anybody could point me any clue about an Open Source or licensed
switchboard for my users?
ARI or FOP is not enought for my users.
Thanks in advance.
VoipCrazy
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Ok got it up and running. In the case for Qwest with NFAS they reserve what
they call Interface ID 1 for the circuit with the backup d channel. In
our case we only have two circuits with a single d channel. The real key
was realizing the logical span number in the spanmap translated into
in
1.2 you can't bring up PRI outside asterisk, since the PRI (I'm
assuming layer 2+) part loads with Asterisk.
On Sat, Jun 12, 2010 at 10:51 AM, Voip Asterisk aster...@wideideas.com
wrote:
Ya i'm not even to the asterisk part yet. I'm still trying to get dahdi
to
bring up the PRIs without
=yes
group=1
channel = 1-23
group=2
channel = 25-48
If anyone could let me know what I should be doing next. I'm sure my issue
is:
Status: Provisioned, Down, Active
specifically the Down part.
Thanks
On Sat, Jun 12, 2010 at 9:04 AM, Voip Asterisk aster...@wideideas.comwrote:
BTW these were
Ya 99% sure that isn't it since they were just pulled working off an AS5300
On Sun, Jun 13, 2010 at 4:27 AM, Doug Lytle supp...@drdos.info wrote:
Voip Asterisk wrote:
Status: Provisioned, Down, Active
specifically the Down part.
In my experience that usually means the provider
Hi,
I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm
using an OpenVox D410E and the drivers are loaded. My system.conf looks
like this:
# Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 B8ZS/ESF RED
span=1,2,0,esf,b8zs
bchan=1-24
# Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2
Ya i'm not even to the asterisk part yet. I'm still trying to get dahdi to
bring up the PRIs without alarms.
On Sat, Jun 12, 2010 at 4:58 AM, Doug Lytle supp...@drdos.info wrote:
Voip Asterisk wrote:
Hi,
I'm trying to bring up two PRIs from qwest with asterisk and dahdi.
I'm using
settings
on dahdi/asterisk?
The line card used in the cisco was the standard 4 port T1 PRI card.
Thanks
On Sat, Jun 12, 2010 at 7:51 AM, Voip Asterisk aster...@wideideas.comwrote:
Ya i'm not even to the asterisk part yet. I'm still trying to get dahdi to
bring up the PRIs without alarms.
On Sat
Hello,
I need to limit the RTP ports used by an asterisk in a client,
Actualy the range defined is from 1 to 2 udp ports.
If I only have 10 local sip extension ¿how many ports/range should I
set up in /etc/asterisk/rtp.conf?
Which is the way to calculate the rtp ports needed on an
Hello all,
I've to deploy about 200 snom320 phones on a instalation.
Do you know any knid of tool to help me with this amount of phones?
I'm thinking in a provisioning tool which I use for setting up the
phones.
Any clue would be welcomed.
Thanks.
Voip-Crazy
Hello,
We're trying to receive G.711 (aLaw) faxes on the asterisk and convert
them to tif. With T.38, we have several issues, so we are trying to use
G.711, since the gateway is located in the same LAN, so there's no
bandwidth/packet-lose issue.
We also use on the same Asterisk Real-Time
: Unknown or unavailable item requested: 't38passthrough'
-- Executing GotoIf(SIP/Proxy-0005, 0?5:T38,1)
-- Goto (fax,T38,1)
Now, referring to the error above, I see (in voip-info.org) that
t38passthrough is an R/O variable and not an R/W, but in any case, I got
0 as a result, so
Now, referring to the error above, I see (in voip-info.org) that
t38passthrough is an R/O variable and not an R/W, but in any case, I got
0 as a result, so it should have been OK, and it's not, as ReceiveFAX
still sends a T.38 reINVITE. If I can't modify it, what should I do
-users] Fax throughput - Asterisk 1.6.1.9
From: Cyprus VoIP voi...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Friday, 04 December, 2009 18:21:59
It's probably because you are using 1.6.1.9; that release (and older)
had a 'feature
Cyprus VoIP wrote:
This is the reINVITE SDP received from the SIP Proxy:
---
Content-Type: application/sdp
Content-Length: 353
v=0
o=root 30427 30428 IN IP4 194.98.xxx.xxx
s=session
c=IN IP4 194.98.xxx.xxx
t=0 0
m=image 17548 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate
Joseph,
You may want to try RPA-2E1S1O from www.broad-tel.com from China. It
provides real FXO port that registers with Asterisk.
David
On Sat, Dec 12, 2009 at 1:37 AM, Joseph syscon...@gmail.com wrote:
I'm looking for a reliable ATA FXO/FXS adapter.
Linksys 3102 - a lot of echo problem + two
VoipFaxMaxRate = 5
; The span over which parity is calculated for FEC in a UDPTL packet
;
udptlfecspan = 3
;
; Some VoIP providers will only accept an offer with an even-numbered
; UDPTL port. Set this option so that Asterisk will only attempt to use
; even-numbered ports when negotiating T.38. Default
We're trying to receive faxes on the Asterisk server, but for the time
being T.38 negotiation fails.
The SDP that the Asterisk reINVITE sends contains these lines:
--
m=image 4968 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval
Hello,
We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would
like to use the sip,extensions and voicemail in realtime mode.
Where can we find the database tables structure for these versions?
Thanks,
Andreas
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To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Thursday, 10 December, 2009 05:26:07
On Dec 9, 2009, at 10:15 PM, Cyprus VoIP wrote:
Hello,
We just installed a new 1.6.1.11 system + 1.6.1.2 addons and we would
like to use the sip,extensions
Cyprus VoIP wrote:
Thank you for your answer. The 'internal extension' is indeed a T.38
capable device that works perfectly when connected directly to the
Proxy/ITSP.
As you said, the key to debugging/resolving this issue is the logger. I
wasn't aware of this file. this is what I have
Set 'canreinvite=no' on all applicable peers?
I tried with yes and no. No difference. I'm almost certain it's related
to the Keeping RTP active during T.38 session issue.
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Cyprus VoIP wrote:
So, I enabled the full logger, and the strange thing I see is this message:
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session
It seems that this might be the reason Asterisk initiates a reINVITE
with voice codecs, after connecting the 2 parties
It's probably because you are using 1.6.1.9; that release (and older)
had a 'feature' that allowed automatic switching back to audio from T.38
if one of the endpoints sent an audio packet. It turns out that wasn't a
good idea, and it's been removed... but in later versions. You'll have
to
Hello,
We are trying to send faxes by T.38 protocol to a remote SIP proxy from
a local extension. The local extension sends the INVITE, Asterisk sends
the call to the Proxy the call is connected with a regular audio codec.
After a few seconds the remote proxy sends an INVITE with UDPTL and the
We set t38pt_udptl=yes in sip.conf and allowed all the codecs to the
local extension and remote Proxy, but it still forces the call to go
back to a voice call.
Define 'internal extension'. Is this a T.38-capable device? If not,
Asterisk doesn't support TDM-to-T.38 FAX relay (yet). If it
Hello,
I tried to install Asterisk + Asterisk addons + FreePBX (latest versions
of all), but in the FreePBX screen, I don't have the option to set ring
groups and IVRs
.
Can anyone tell me what I'm doing wrong?
Thanks,
Andreas
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I tried to install Asterisk + Asterisk addons + FreePBX (latest
versions
of all), but in the FreePBX screen, I don't have the option to set
ring
groups and IVRs
Can anyone tell me what I'm doing wrong?
You are not posting on the FreePBX forums? ;)
I figured Asterisk-Users would
The problem is that the online module update is not working for me
(Cannot connect to online repository (mirror.freepbx.org). Online
modules are not available.) and I couldn't find online a working
solution :-(
DNS/Gateway ok on server?
Yes. The problem is with the FreePBX modules. I
From: Cyprus VoIP voi...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Saturday, 03 October, 2009 09:28:20
What does your musiconhold.conf look like?
[general]
[default]
mode=files
directory=/var/lib/asterisk/moh
Hello all,
I want to instal a Billing solution in the same asterisk's box. I have
browse for ast2bill asterisk billing, astercc, and more, bu ti do not
know which will be the best for me.
The only things i need, are,
- Postpaid and prepaid applications.
True CDR,
that asterisk one, With suport for transfers
- I do not need support for reseller
- Billing for Voip, PSTN trunks
I need a light app. I'm not searching a heavy app. with a lots of
modules and applicacions. I need a ligth application for a soho and
its needs.
Any one are using
What does your musiconhold.conf look like?
[general]
[default]
mode=files
directory=/var/lib/asterisk/moh
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now:
-users@lists.digium.com
Date: Wednesday, 30 September, 2009 15:27:28
On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote:
snip
You see the wav files but do you see the files encoded for the codecs
you are using?
There's only one wav file there. No encoded files, but on asterisk 1.2
we have
What is the output of moh files show CLI command ?
pbx*CLI moh show files
Class: default
File: /var/lib/asterisk/moh/manolo_camp-morning_coffee
File: /var/lib/asterisk/moh/macroform-the_simplicity
File: /var/lib/asterisk/moh/macroform-robot_dity
File:
Hello,
Has anyone encountered that when Asterisk sends RTCP messages, it stops
sending RTP packets until it gets an answer?
Can that be fixed?
Thanks.
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AstriCon 2009 -
/fix it, your help would be HIGHLY
appreciated. We're really stuck.
Thank you all in advance.
Original Message
Subject: Music On Hold
From: Cyprus VoIP voi...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Tuesday, 29
I'm afraid I can't be much help as I am both a newbie and it works just
fine for me on 1.6.1.6. Of course, mine was a fresh installation.
Thanks for your help, John. Mine is also a fresh installation, but now
at least I know it's not a version issue.
Is there anything in the logs to give
Hello,
We need help in debugging Music On Hold on our Asterisk 1.6.1.6
From the SIP debug, I see that an extension sends an INVITE of the call
to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but
I don't see in the console any reference to the call being placed on hold.
Hi,
Is there anyone there that installed successfully the CRI package and
manages to play the calls listed in the call monitor page?
Regards.
Original Message
Subject: Re: [asterisk-users] Crystal Recording Interface
From: Cyprus VoIP voi...@gmail.com
To: Asterisk Users
Hello,
I see that there's 1.6.0.x and 1.6.1.y versions of Asterisk.
Is there a clear table that describes the features and/or differences
between them?
Are both stable enough?
Is T.38 Fax supported on both? If yes, which spandsp is supported? I saw
on voip-info.org that version 6
Hello all,
I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon, but
without success.
Is there a proper online manual that describes all the steps to follow
and debugging/monitoring information?
When I type in the CLI module show, cdr_addon_mysql.so is not listed,
although in
Message
Subject: Re: [asterisk-users] Need help - CDR MySQL
From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 17:17:59
On Sunday 30 August 2009 08:30:54 Cyprus VoIP
, 2009, at 11:18 AM, Cyprus VoIP voi...@gmail.com wrote:
Thanks. I found out that the module didn't load:
[Aug 30 20:35:59] WARNING[31906]: loader.c:371 load_dynamic_module:
Error loading module 'cdr_addon_mysql.so':
/usr/lib/asterisk/modules/cdr_addon_mysql.so: cannot open shared
object
...@hh174.be
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 18:58:48
yum search mysql client
yum install 'TheClientYumHasReturnedForYourSystem'
Olivier
Cyprus VoIP a crit:
I think that the missing component
-Commercial Discussion
asterisk-users@lists.digium.com
Date: Sunday, 30 August, 2009 19:27:46
Cyprus VoIP wrote:
I think that the missing component is mysqlclient, but when i yum
update mysql, it does nothing.
You need to make sure that mysql-devel is installed and then re-compile
add-ons
Hello all,
I download from Tikal's site the Crystal Recording Interface and
installed it on my Asterisk server, but there's no reference in the
installation instructions there regarding the necessary settings on the
Asterisk itself.
Is anyone using it? Any detailed explanation on the
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cyprus VoIP
Sent: Friday, August 28, 2009 9:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Crystal Recording Interface
Hello all,
I download from Tikal's site the Crystal
I just plug the junper in NT mode with no success.
VoipCrazy
2009/8/15 Paul Hales pdha...@optusnet.com.au:
Use a standard network cable - but you have to activate the 'terminate'
jumper on the NT end.
- Also, the new BRI stuff in dahdi is much easier to work with than misdn.
PaulH
voip
Hello all,
I'm trying to conect two asterisk servers using two B410p Digium
cards. One card on each server. I just setting up the first BRI port
on server A as nt_ptp and the first BRI port on server B as te_ptp.
I use an ethernet wire to connect the first port of server A (nt_ptp)
with the first
Message
Subject: Re: [asterisk-users] INVITE Privacy Information
From: Philipp Kempgen philipp.kemp...@amooma.de
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Monday, 27 July, 2009 17:16:45
Cyprus VoIP schrieb:
I would like to use Asterisk
Information
From: Philipp Kempgen philipp.kemp...@amooma.de
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Tuesday, 28 July, 2009 14:10:55
Cyprus VoIP schrieb:
I ran into this problem: When I change the CALLERID(num and name) to
anonymous
Hello all,
I would like to use Asterisk to add/modify SIP headers in the INVITE
message, to include Privacy information, if the INVITE includes a *67
prefix (or another predefined prefix).
That's an example of the INVITE I get:
/INVITE sip:*6700112233...@192.168.1.100 SIP/2.0
From:
Hello,
I would like to add SIP headers to the REGISTER messages Asterisk (1.6)
sends to an external proxy.
Also, I want to be able to reorder the lines.
Is it possible?
If yes, how?
Thanks.
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Hello list,
I have an asterisk / hylafax / iaxmodem configured in one machine. All
is working nicely. Now I need the fax to be print when arriving.
¿Anybody have this feature implementing in their systems?
¿How is the best way to get that?
Any clue will be welcomed.
Thanks.
VoipCrazy
Hello all,
I need to configure an application which let me to call from a web page.
Someone has experience using apps to make webcalls?
Which software do you use?
Thanks.
VoipCrazy.
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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
Hello list,
Does anybody know any free WebCall solution to let our customer call
us directly via our web site?
Any clue will be welcomed.
Thanks.
VoipCrazy
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asterisk-users
Hello list,
I have got an asterisk box installed working ok with an b410p card to
make and receive isdn calls.
All works ok, but when a call is answer and the person starts to
speak, always I can ear a beep during the call. This beep is ear
some times in about 30 seconds between each beep.
When the asterisk a queue reset their counters?
I 'm talking about this kind of info in asterisk console.
show queue 600
600 has 0 calls (max unlimited) in 'ringall' strategy (4s
holdtime), W:0, C:14, A:8, SL:0.0% within 0s
I just say that because I have a queue with strategy Fewest
Dear List:
I need to make a sip phone (spa942) answer a call but the phone must
no ring. The user only has to show the callerId on the phone screen
without any sound.
How could I make that in asterisk? I tried to use Sip headers but I do
not know how must I say the phone don't ring when
According to Your description this is a phone problem.
Asterisk behaves as its expected.
post your dundi.conf to dig more in to this.
regards
rama
On Wed, Sep 24, 2008 at 9:52 PM, ronald ramos [EMAIL PROTECTED]wrote:
hi,
when a user register on my asterisk i can see it adding Noop for that
to do it myself so I generated the two keys (pri and pub) for
each server with their own hostname then I copied:
- .121 keys to the other two servers (.137 and .204)
- .137 keys to .121
- .204 keys to .121
Let me know how if it works.
Giorgio Incantalupo
technocrat voip wrote:
Hello All
Hello All,
Iam trying to achive a simple load balancing with dundi.
Here i have three asterisk boxes like below.
*.*.*.121 which is the dundi server
*.*.*.137 A Peer which has the 1000 phone registerd to it
*.*.*.204 B Peer which has the 200 phone registered to it.
The expected behavior of
on, do this in the first of sip.conf file
Best Regards
On Mon, Sep 1, 2008 at 11:32 AM, voip crazy [EMAIL PROTECTED] wrote:
Hatem,
I cannot understan exactly what you told me.
Could you try to explain that in other words. Better if you could post
an example of this SIP trunk.
thanks
Hello list,
I have an asterisk instalation with a bad internet connection cause
this connection is down sometimes.
When the connection is down and asterisk cannot get internet
connection. All the extensions log out from the asterisk machine, and
nobody can make any call.
¿Why if internet
issues with your
router? Can you reach all the boxes in your lan while you are
experiencing this downtime?
voip crazy wrote:
When I say extensions, I say extensions in the lan not in wan
Thanks.
VoipCrazy.
2008/9/1 Igor Hernandez [EMAIL PROTECTED]:
Hello,
By people do you mean people
When I say extensions, I say extensions in the lan not in wan
Thanks.
VoipCrazy.
2008/9/1 Igor Hernandez [EMAIL PROTECTED]:
Hello,
By people do you mean people in the lan or external users?
Regards,
--
Igor Hernandez
Escape Communications
http://www.escapetel.com
voip crazy wrote
Hatem,
I cannot understan exactly what you told me.
Could you try to explain that in other words. Better if you could post
an example of this SIP trunk.
thanks in advance.
Voip Crazy
2008/9/1 hatem moiz [EMAIL PROTECTED]:
Asterisk is looking for a SIP trunk if you have recorded the usage
Hello users,
Can you share with me your experiences with Asterisk 1.6? Is it stable
enough for commercial service?
Thanks.
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Hello list,
How could I limit the outgoing calls for one trunks easily?
Thanks
VoipCrazy
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
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Hello all,
A client of us, is thinking to migrate their actual PBX to a Cisco
CallManager. We want to sell him an asterisk box to complement the
Cisco PBX.
I think to use asterisk as a Voicemail server (Replazing the Cisco Unity)
Has asterisk all the functionalities to replace a CIsco Unity
Hello all,
I need to install asterisk for 900 sip users with 2 PRI ports.
It is posible to handle this number of calls/extensions with only one
asterisk machine?
Which is the best way to install that? two asterisk with openser. One
asterisk with openser .
Is it necesary run a SER server on
Maybe 400 calls at one time. By the momento there aren`t voip trunks
maybe in the future.
About cluster, Which cluster solution will could be good option?
Which solution could I use to do load balancing between two asterisk machines?
Thanks again.
Voipcrazy
2008/7/9 Tom Moore [EMAIL PROTECTED
Hello friends,
I am using the asterisk-1.6.2 , i use the gui also.
I use the polycom provisioning.
Now my requirement is to allow the phone to upload the log files etc to the
asterisk machine.
As i see now when it queries to upload the file
like below
T 10.231.109.206:1037 - 10.231.109.59:80
Hello,
I want to create an script which remove all the old voicemail messages.
I make a simple Bash script to delete all the new messages for the
extension 100. Something like,
rm /var/spool/asterisk/voicemail/defaul/100/INBOX
Should I update any index file or something after reemove them?
Hello all,
Some one is using asterisk and queuemetrics connected via astmanproxy?
How about your experience?
Which proxy do you use in this kind of connection?
In my instalation asterisk and Queuemetrics are installed on diferent
machines and I want to avoid manager problems
Thanks in advance.
Hello all,
Someone knows any softphone which accept messages using sipsak?
I just tried X-Lite and portsip without success
Thanks
Voipcrazy.
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AstriCon 2008 - September 22 - 25
Hello all,
I have an asterisk PBX working perfectly, and the transfers between
extensions, works ok. The problem, when I receive a call from the line
connected to the TE12Xp, and I try to transfer it, the calls hangs up.
I have other analog lines and I can tranfer all the without problems.
I've
More info about the problem.
This occurs, when I try to transfer using the *2 funcionality into aterisk
Thanks
2008/6/16 voip crazy [EMAIL PROTECTED]:
Hello all,
I have an asterisk PBX working perfectly, and the transfers between
extensions, works ok. The problem, when I receive a call
I need to execute an action after a call is hangup. I just see the
command Dial has an option for that, the g option.
I configure the dial command as
exten = s,n,Dial(SIP/100,100,Ttg)
How should I add the line which the command will be executed after the
dial command in this example?
I don`t
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