I have a strange problem: if I put on hold an incoming call from my Digium
TE110P, I can't resume it and the person at the phone continues to hear MOH
until the line falls.
My TE110P is connected with an italian E1 NT.
If I put on hold a call on a SIP channel I can resume it without any
14 mar 2006 kl. 01.45 skrev Steve Kennedy:
On Mon, Mar 13, 2006 at 07:38:01PM -0500, Watkins, Bradley wrote:
That depends on what you mean by default. The supplied sample
extensions.conf contains the priorityjumping=no by default, but if
this
parameter is absent then the default is to
Dan,
what is so wrong with the snom360 ? I now your wiki website, but as far as I
can see, nearly all major issues are resolved. Meanwhile we have the version
5 branch much more stabilized, see beta 5.5:
http://www.snom.com/wiki/index.php/Beta_Firmware
and if you don't like to use a beta,
Hi,
OK, that will enable the auto generation of a context but as the new
context won't have a switch statement it doesn't help with this
problem... I may try writing a default switch if no matching context
found type patch.
Peter.
On Mon, 2006-03-13 at 20:51 +0200, Benchev wrote:
I was
How can I forward my offcial sipgate number to different users, I would
like to know if it is possible to append a local user number to my
official number when dialing, then in this way it could be forwarded
using the suffixe local user number.The prefixe number would be the
official sipgate
When incoming DATA call arrive on ISDN BRI, asterisk (zaphfc) recognise
type of call, but answering anyway (playing IVR messages, ringing
phones, etc...)
How to stop that? I want that only VOICE calls are answered, and
DATA/FAX to be ignored.
(I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f,
Yes it does display caller id as callingnumber@ip of calling party but
that does not interfere with me hitting dial from missed calls. Seems the
Cisco phone sends the sip INVITE as callingnumber@ip of calling party
rather than callingnumber@ip address of defaultproxyserver but asterisk
Maybe I have something strange in my dial plan but I have no problem just
hitting dial from missed calls under 8.2.
Chris
- Original Message -
From: Aaron Daniel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
I have found in the past that using the resample -ql option gives better
results.
Chris
- Original Message -
From: lenz [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 14, 2006 7:12 AM
Subject: Re:
Reply to self:
Last week I had some time to figure out a workaround for the CDR logging
problem. I used an AGI-script together with de Mysql CLI application. It is
far from perfect, and I want to spend some more time to figure out a better
way, but this seems to be working OK on my testmachine:
My asterisk system seems to have problems detecting hangups. I am
getting a LOT of voicemails with dialtone or silence.
I am using an external gateway (wellgate 3701a) and don't have zaptel
at all.
I think your 3701a don't understand hangup tone (as our 3802 did and
keep line busy after
I was able to install Asterisk and Asterisk-addons and use them
successfully. But I have a problem now, I have many contexts and it
looks like Asterisk is unable to find the context given directly in
Mysql DB unless I specify it in Extensions.conf to switch it to
My systems work perfectly with 8.2, hit dial from the missed calls menu
and the call is placed exactly as expected.
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Stenton
Sent: 14 March 2006 09:47
To: Asterisk Users Mailing List -
Jerry Geis wrote:
I was searching on voip-info.org for saydigits.
I see no indication it is not valid in 1.2.4 asterisk.
however, when trying to use it I get and error no application saydigits.
what is the correct way to echo back digits in asterisk 1.2.4?
I tried say digits 123 and saydigits
Looks like the original question posed to the OP had to do with physical
wiring, as in red/green equals line #1 and yellow-black equals line #2.
James Harper wrote:
Definitely one line per FXO port, but the wording of the original poster
was two numbers, not two lines, and while it may not be
Right saw that. But I'm trying to get away from using CVS-HEAD :)
Is the jitterbuffer patch PURELY 1.2.5 with the patch in place?
On 3/14/06, Olle E Johansson [EMAIL PROTECTED] wrote:
13 mar 2006 kl. 21.59 skrev Matt:
Hi,
I really want to start using 1.2.5, but I also really need to
Chris, you may have a different and simpler setup. Internal calls work
fine here, since the proxy server on the CallerID is the same proxy
server used for all internal users. I was referring to calls that
originate outside of the enterprise. I should have been more clear.
Omar
On 3/14/06,
I found that the Fax detect delay in the extentions.conf was causing my
system to have a delay
[from-pstn-reghours]
exten = s,1,GotoIf($[${FAX_RX} = disabled]?from-pstn-reghours-nofax,s,1:2)
; if fax detection is disabled, then jump to from-pstn-nofax - else continue
exten = s,2,Answer
exten =
Anyone knows if the SC430, based on the Intel E7230 chipset, is
compatible with the Digium cards? I've tried the compatibility page on
digium's website. It seems like they've pulled the old compatibility
list, now the links on the page only point back to the product pages.
Over here, Dell is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tue, 14 Mar 2006, Omar A. Sabek wrote:
Chris, you may have a different and simpler setup. Internal calls work
fine here, since the proxy server on the CallerID is the same proxy
server used for all internal users. I was referring to calls that
We only had the problem when the call was redirected from one server to
another. So if a phone was called from another phone on the server, the
called worked perfectly, but if it was redirected from another server,
we got the proxy added to the end. Doesn't help when you're trying to
make
Sangoma are about to release a 2-port card I believe, but I have not
heard of a 1-port unit. You would need to buy an external device,
which will probably raise to cost so close to the 2-port solution that
you may as well use that instead.
Regards,
Steve
On 3/9/06, Avi Miller [EMAIL PROTECTED]
14 mar 2006 kl. 13.35 skrev Matt:
Right saw that. But I'm trying to get away from using CVS-HEAD :)
We all are. Every developer have switched from CVS to Subversion :-)
This is not the development branch, but the release branch code,
which we use to create the 1.2.x releases.
The
Hi all,
I've bought a TE110P, and received it today. So i decided to install
[EMAIL PROTECTED] 2.7 with this card.
In the past i had experiencies with X100P (clone card) and it never
take me so long to reboot the machine
Machine:
P4- 2,8Ghz 1GRAM
TE110P
What could be wrong?
Best regards,
Anyone knows if the SC430, based on the Intel E7230 chipset, is
compatible with the Digium cards? I've tried the compatibility page on
digium's website. It seems like they've pulled the old compatibility
list, now the links on the page only point back to the product pages.
Over here, Dell is
On Monday 13 March 2006 23:16, Anton Krall wrote:
Might be good for faxing though
Doubtful. Faxes are designed to work within g711 limits. I personally have
been faxing through Asterisk (Canon and Xerox fax machines, the most
notorious for being fickle) for well over a year now. It generally
Is there a way to connect a phone line to
another line that is in the offhook state?
The Dial() application evidently needs to call
the other line (onhook state or a busy signal given), I would like the other
line to be already offhook and the phone line then gets connected.
Thanks
I not understand why my asterisk send the tone of pound key (#) only
when i click twice time.
I deactivate the transfer function.
Matteo
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Hi,
i'm trying without success to change the dst (destination) entry of the
cdr. I'm using the following:
exten=_2006234500254.,2,SetVar(CDR(dst)=${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
I want to record into the cdr only the called number, but in the
The jitterbuffer branch is based on svn trunk (the same as the old
CVS HEAD)
The jitterbuffer-1.2 branch is based on the 1.2 branch HEAD
(meaning latest 1.2 version code).
Ok... so if I pull the 'jitterbuffer' branch I would get 'svn trunk code'.
But if I pull 'jitterbuffer-1.2' I get the
Hi,
I recently purchased a brand new 7905g with his SIP firmware (licensed).
Now, I want to play a little with chan-sccp, but I'm unable to find the
appropriate firmware for my phone. I know that I must get it directly
from cisco, but before purchasing it will be very good for me to try a
bit;
The sipura 2100 does work good with a AS5300
Zoa wrote:
Does anybody know what devices really support t.38 ? I've seen a few
claiming they do on the box, but most do not seem to support it at all.
Zoa.
Kristian Kielhofner wrote:
Olle E Johansson wrote:
Friends in the Asterisk.org
Sean Cook wrote:
Yes you do need unixODBC before you compile asterisk. Once you have
installed unixODBC , asterisk will compile and offer you the following
modules:
cdr_odbc.so
res_config_odbc.so
res_odbc.so
res_odbc.conf and cdr_odbc.conf are the related config files...
Now, I have
I'm still having problems with ser and asterisk on the same public
server.
Could anybody send me a tarball of their ser.cfg and sip.conf off-list,
so I can do a sanity check against my files?
Much appreciated.
Bart...
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Is anyone using realtime sip for friends (ie phones) with multiple Asterisk
boxes all pointing to the one central MySQL database? Does it work? Are phones
that are registered to the database from Asterisk box able to reach phones
registered to the database from another Asterisk box?
Doug.
i'm trying without success to change the dst (destination) entry of the
cdr. I'm using the following:
exten=_2006234500254.,2,SetVar(CDR(dst)=${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
I want to record into the cdr only the called number, but in the cdr
Matt,
Without getting into a phone war...
What phones or headsets or softphones do you use with your installation?
Thanks
Adam
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell
Sent: Friday, March 10, 2006 1:51 PM
To: Asterisk Users Mailing
If you go into the BIOS and disable all unneeded devices (serial, parallel,
USB, floppy, etc) then you shouldn't have any problem. I have one in a 15
user setup that is working fine.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Leo Ann
We were using this setup for a while (well, it was using odbc, but same
concept). What we did was configured the phones to register with all
the servers basically, so each phone was reachable by each server, and
if a phone didn't register with a server for some reason, we have
mechanisms in
Sorry for the trivial question I did
The answer is:
Only install linux kernel-default in Yast Software Management
Andrea
[EMAIL PROTECTED]
Title: [Asterisk-Users] Clustering "NEW THREAD", Almost Working
Now, Iknow what
you guys been talking about. It is likeDSN forsip phones, not really
clustering. I original thought that you guys want to setup some thing that can
fail over to a different sip server if the server running the
Hi Doug,We use Realtime SIP via a central MySQL database (2 actually in Master Master config) but registration is only available on the box to which the client has registered. Clients can register with any database and the table does get updated with some registration information (ip address,
Hi,
Using openser 1.1.0-dev8 as a registrar/proxy in from of Asterisk.
Recently I have been getting errors from Asterisk due to corrupted From:
headers, which appear to be caused by uac_replace. Here is a section of
the debug log:
Mar 14 15:12:00 www1 /usr/sbin/openser[7933]:
I may be way behind here, but I see that digium redesigned their site.
I cannot find the mailing list search screen.
I have found the mailman list page, but that doesn't have have a nice
search ability.
Do I need to just rely on google and other generic search engines or is
there a search on the
At 11:32 AM 3/13/2006, John Daragon wrote:
Phil Freed wrote:
I'm afraid that I am at a loss here. I am new to Asterisk, and have
successfully set up SIP. But I cannot get my FXS card working, and I'm
not sure what else I can try.
# modprobe wcfxo
I prefer the google groups search
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael George
Sent: Tuesday, March 14, 2006 9:49 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] digium.com redesign
I may be way behind here, but I see
Hi Michael -
I may be way behind here, but I see that digium redesigned their site.
I cannot find the mailing list search screen.
I don't believe there has ever been a search screen.
Do I need to just rely on google and other generic search engines or is
there a search on the digium site?
I have done this but I still get choppy sound and echo on some callsthanksGiovanni Miano [EMAIL PROTECTED] wrote: Of course,Echo is 2 types: electric and ambiental.If u gain rx o tx more than you need, its return in recive and gen echoTry to decrase value, try to set 0 or .. in samecase -1
Just poking this topic as it seems to have been ignored. I still am not
clear as to how/where this script is broken.
If I read this correctly the syntax in column two is the current best
practice for AstDB. It, unless I've missed something below is what I
have used in my script.
-Original
Need 2 full-time Asterisk Guru's needed in Phoenix area right away.
Knowledge in some of following also needed:
Php
Perl
MySQL
PostGres
C++
Visual Basic
HTML
Photo Shop
Email resume off list - Will interview this week.
Kyle
--
CONFIDENTIALITY NOTICE: This message,
I apparently have a dead FXS module. Is there any kind of test I can
run on it (on a live system) to determine if its good or bad? The
telephone has no dialtone, gets no calls. It has been working for
several months, and just quit yesterday. Thanks!
Found a Wildcard TDM: Wildcard TDM400P
I tried it, it didn't work, and I tried without success the following
exten=_2006234500254.,2,SetVar(destination = ${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/[EMAIL PROTECTED],60,tr)
Any help will be welcome
thanks
Benchev wrote:
i'm trying without success to change the dst
Based only on what I see below (from previous posts), it sounds like you
have two separate issues going on: 1) echo, and, 2) choppy sound. Those
should be analyzed as two problems (not one).
You will find plenty of posts in the archives relative to both. In
general terms, the choppy audio
On Tuesday 14 March 2006 17:15, Benchev wrote:
i'm trying without success to change the dst (destination) entry of the
cdr. I'm using the following:
exten=_2006234500254.,2,SetVar(CDR(dst)=${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
I want to record
Hi,
We have two Asterisk servers connected over IAX, with very limited bandwidth
256Kbs.
When we make calls between these two Asterisk servers the sound is very
choppy, no matter whether we use jitter buffer or not.
However, when we make calls using Skype, the sound is perfect.
Can anyone help
Best bet... call digium support. The module should be under warranty
(two years), and they will be able to tell you how to test if the notes
below aren't already enough.
Jimmy wrote:
I apparently have a dead FXS module. Is there any kind of test I can
run on it (on a live system) to
i'm trying without success to change the dst (destination) entry of the
cdr. I'm using the following:
exten=_2006234500254.,2,SetVar(CDR(dst)=${EXTEN:10})
exten=_2006234500254.,3,Dial(OH323/${EXTEN:[EMAIL PROTECTED],60,tr)
I want to record into the cdr only the called number, but in the
thanks for the info.it is not sharing an irq: 0: 59840409 59803082 IO-APIC-edge timer 8: 1 0 IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 11: 0 0 IO-APIC-level ohci_hcd:usb1 14: 2141851 2143209 IO-APIC-edge ide0 177: 111558 111273 IO-APIC-level aic7xxx 185:
15 0
I have 3 POTS lines that I want to use with Asterisk, I am looking at
prices for FXO cards and the cards with echo cancellation are really
pricey... is echo cancellation really worth it for a 3 or 4 line
system? Will I notice a difference without the echo cancellation?
Thanks
Keith Schmidt
Andrew,
Don't know if this helps your or not, but it seems like you have one too
many {} in your set statement...
You have: Set($ { DB( forward/${CALLERIDNUM} ) = ${FORWARD} } )
Try: Set(DB(forward/${CALLERIDNUM}=${FORWARD}))
- Jason
-Original Message-
From: [EMAIL PROTECTED]
Irvine California, Heritage Park Library on the
corner of Yale and Walnut. The Walnut is just south of the 5 fwy and Yale is
between the Culver and Jeffery offramps. Meeting will run from 6 - 9pm. This
week will feature a review of the SNOM 320, a demo of SIPX, some book giveaways
courtesy
Hi,
I have an issue which is kind of a catch 22 situation. I had outgoing calls
to my new PSTN provider working perfectly. Then I started focussing on
incoming calls. It seems that I can solve an error which gets my incoming
calls working but that in turns means my outgoing calls don't work. -
Is it
just me or is the voip-info web site down right now? Jeez that web site is
flaky.
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Hello. I am following the directions in your legal disclaimer because
I received a copy of your message, yet the message was not addressed
to me. My e-mail address is [EMAIL PROTECTED], but the message I
received was addressed to [EMAIL PROTECTED]
I would hate to see your confidential electronic
On Tue, 2006-03-14 at 10:21 -0700, Douglas Garstang wrote:
Is it just me or is the voip-info web site down right now? Jeez that
web site is flaky.
Is it just me or was this message in HTML? Jeez some people never learn.
Dave Cotton [EMAIL PROTECTED]
Hi,
We are trying to use attended transfer with Asterisk 1.2.5, but when we
do the transfer and dial the new number, it times out after 3 rings and
then the callee is put back to the original agent.
Where can I adjust the timeout which applies to the number we are
transferring to? I have
Hello to all
Does someone knows how to force the Cisco IP phones (7940 and 7960) to
reboot weekly or monthly?
I think this would be useful because sometimes we change the
configuration settings in the TFTP, but the phone just check the TFTP
when he restarts...
Thanks
João
Oh, I'm sorry. I must have missed the previous message where you specifically
informed me not to use HTML.
-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 14, 2006 10:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Mar 14, 2006, at 9:04 AM, Keith Schmidt wrote:
I have 3 POTS lines that I want to use with Asterisk, I am looking at
prices for FXO cards and the cards with echo cancellation are really
pricey... is echo cancellation really worth it for a 3 or 4 line
system? Will I notice a difference
Yes, SIP realtime is working with multiple * servers all accessing the same
MySQL database, add a sip phone in the database and the phone can register with
any server without the need to configure any server, just add the phone in the
database, petty cool.
JR
--
We send a notify message with the check sync event type. Not pretty but
it works.
Joao Pereira wrote:
Hello to all
Does someone knows how to force the Cisco IP phones (7940 and 7960) to
reboot weekly or monthly?
I think this would be useful because sometimes we change the
configuration
On Tue, 14 Mar 2006 14:32:02 +0100
Olle E Johansson [EMAIL PROTECTED] wrote:
14 mar 2006 kl. 13.35 skrev Matt:
Right saw that. But I'm trying to get away from using
CVS-HEAD :)
We all are. Every developer have switched from CVS to
Subversion :-)
This is not the development branch, but
It really depends on the number of phones you're wanting to reboot.
Whenever we do a reconfiguration of our phones, I have a script that
runs that night that pulls all the names from the db that are cisco
phones, and does a sip notify cisco-check-cfg exten in asterisk, which
notifies the phone
Well... the next step (for me anyway) would be to use Ethereal on the
asterisk nic interface to ensure the sip/rtp pkts are reasonable (eg, no
dropouts). If those pkts flow consistently in both directions, then
there must be something impacting the wctdm interface.
Do sip to sip calls sound
Hello,
We use mostly Channelbanks and cheap analog phones with nice headsets.
Much cheaper in the long run and much easier/faster to replace the
phones. We also use Sipura ATA adapters with cheap analog phones and
nice headsets, and for our remote/external agents we use Firefly
third-party(free
The best way to set gains would be to use ztmonitor (located in
/usr/src/zaptel). Make a call and note your channel number. Run
/usr/src/zaptel/ztmonitor channel number -v from a telnet session. check
to see if your levels are too high or too low and adjust your zapata.conf
accordingly. I
Does anyone know if their are rules that this list is supposed to be
following? It doesn't appear to be moderated, so I realize that such
rules would be self-enforced, but it still might be good to agree on
some. Likewise, we could agree on none. That works also. Any
thoughts?
Some
On 14 Mar 2006, at 16:44, Stojan Sljivic - GDS wrote:
Hi,
We have two Asterisk servers connected over IAX, with very limited
bandwidth
256Kbs.
When we make calls between these two Asterisk servers the sound is very
choppy, no matter whether we use jitter buffer or not.
However, when we
Group:
Would it be possible to bridge two channels manually?
Scenario:
user1chanA-AsteriskchanBuser2
user3chanC-Asterisk
At this point, I send reINVITE to user2, and want to
bridge chanB with chanC and then tear down chanA.
My goal here is to make user2 talk with
I think this has been answered before. Try searching in the list
archives, or maybe the wiki :P
Douglas Garstang wrote:
Oh, I'm sorry. I must have missed the previous message where you specifically
informed me not to use HTML.
-Original Message-
From: Dave Cotton [mailto:[EMAIL
I am using the Monitor() application (with
soxmix for combining the audios) and the voice connected to the phone network
is recorded at a lower volume then the voice connected directory to the Zap
analog phone card. How can I get both the audios to be at the same volume on
recording?
Am Tuesday 14 March 2006 18:38 schrieb Barry Flanagan:
Hi,
We are trying to use attended transfer with Asterisk 1.2.5, but when we
do the transfer and dial the new number, it times out after 3 rings and
then the callee is put back to the original agent.
Where can I adjust the timeout which
Are there any step by step instrunctions on how to install drivers and I
guess bristuff for this card?
Just need to use it to handle voice on 2 BRI circuits (UK) then utilise
with Asterisk and some Digium cards handling POTS phones (and some VoIP
out the back).
It's the EICON card stuff and how
Had this working also at some point, but had one killer problem... NAT
issues! Most of our clients are natted, and depending on the router, they
only allow traffic to return from the server that the traffic was sent to.
So the invites coming from other servers were being dropped.
But besides that
http://svn.digium.com/view/asterisk/team/oej/jitterbuffer-1.2/
On 3/14/06, Robert Webb [EMAIL PROTECTED] wrote:
On Tue, 14 Mar 2006 14:32:02 +0100
Olle E Johansson [EMAIL PROTECTED] wrote:
14 mar 2006 kl. 13.35 skrev Matt:
Right saw that. But I'm trying to get away from using
Hi Everyone,
I am using real time asterisk architecture and have placed
the following in sip.conf:
[general]
notifymimetype=text/plain
checkmwi=10
rtcachefriends=yes
but the MWI doesnt work?!
Can anyone give me any pointers as to what the problem could
be?
Thanks
Due to changes at the office, I'm finally getting around to setting up
an AA to deal with incoming calls. One of the big changes is that we're
dropping the old alphanumeric pager and will just send pages to our
phones. I've got the outbound greeting message working in a test
context no
On Tue, 14 Mar 2006 13:44:57 -0500
Matt [EMAIL PROTECTED] wrote:
http://svn.digium.com/view/asterisk/team/oej/jitterbuffer-1.2/
Thank you I was looking directly under asterisk and
not team. :-)
Robert
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Yes. Download the patch from here http://bugs.digium.com/view.php?id=5841
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
JSSent: Tuesday, March 14, 2006 1:15 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] channel
bridging
Group:Would it be possible to
That is a show stopper. However, if your clients are in groups behind
their respected router, you might be able to give them a little linux
app such that this app can PERSONIFY the phones to send a packet to the
respected server.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Does
anyone know if realtime extensions allows extensions in the format
callerid/extension yet? ie the extensions.conf file allows you to
do:
5551212/1000 = exten ...
and it
matches against extension 1000 when the caller id is 5551212. Last time I
checked, realtime didn't support this
All,
How do I get Asterisk to return a 486 SIP response
intentionally?
Thanks,
Jon
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Patrick Friedel wrote:
Obviously extension 2 needs to be changed, right now it just leaves a
message in my mailbox. I'm figuring I'll add a new message that says
Please enter your callback number, followed by the pound sign. and
put that in as a Background() message. The tricky bit that I
Hate to reply to my own post, but figured it out.
Just have to setup the IP Phone to DND.
Thanks,
Jon
- Original Message -
From:
Jon Weisman
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, March 14, 2006 2:36
PM
Subject:
Hi
My name is Jose Manuel Cortes and im developing an
IP telephony project, im going to interconnect a definity prologix PBX with an
asterisk server (i still don't know what kind of cards i'll use digium, sangoma
or voicetronix)trough a E1 connection in order to add ip telephony
tothe
Hi all,
In Brazill, there is a trick to avoid collect calls: if you flash the
line in the first 1000ms, Telco will drop any collect call for you.
Given the R3 signalling here, I have to use LibUnicall. Seems that there
is no Flash command for unicall chanells, just for the Zap ones.
How
I think I've asked this before and think that Matt had said something
about this.
Is there an LCDproc client for Asterisk available and if so how can I
get a copy please.
Thanks
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
___
Central business location has a PRI with a CLEC. Remote offices access
the PRI for all voice traffic via VoIP.
How does one get the telco to report the address of a remote office to
the 911 call center when the call is made from that respective location?
Do you have the right cable?
You need a cross-over T1 cable and NOT a cross-over ethernet cable that
people commonly try. This should satify the electrical requirements and
turn the lights green.
You're on your own with the rest.
I do have a question however; why are you now speaking SIP to
Not to be a smarta**, but you have to ask them to do it. We do the same
thing and it works for us. Depending on the CLEC, they may do it or
they may say no. If they say no, there isn't anything you can do about it.
Hugh L. Johnson wrote:
Central business location has a PRI with a CLEC.
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