On Mon, Dec 11, 2006 at 06:53:19PM +1100, Klaverstyn, David C wrote:
I have since added fxs_ks=1 and channel = 1
This has not fixed the problem. I do notice a warning on the reload of
asterisk.
WARNING[4296]: chan_zap.c:10874 setup_zap: Ignoring signalling
Right. reload of chan_zap will
Thanks for your help.
This is my file.
[channels]
language=au
context=from-pstn
signalling=fxo_ks
;rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
Hi,
I am using Procurve Switches by HP for PoE.
http://www.hp.com/rnd/products/switches/ProCurve_Switch_3500yl-5400zl_Series/overview.htm?jumpid=reg_R1002_USEN
Aside from being a LIFETIME WARRANTY, I found them very easy to configure and
install.
Regards,
Angel
- Original Message
[Sorry I re-send this message as I couldn't see it in the list. I hope it
will not come two times].
Hi everybody.
It is possible to announce the parking position through a paging to a group
of extensions?
I would like that when someone parks a call, some phones will announce with
the
Hi all,
Howto configure asterisk 1.2.13 (debian-base) with support Instant
Messaging, especially using client Xlite v.3.
Thanks
-
This email was sent using Student EEPIS-Webmail.
http://student.eepis-its.edu/
hello everyone,
i have been researching into transnexus (http://www.transnexus.com/)
OSP (open settlement protocol) server. i am really interested in its
routing flextbility and call clearing capabilities. Has anyone
implemented OSP with Asterisk or Cisco voice devices. I would like to
have
Morning,
we have gateways with FXO port registered as SIP endpoint in Asterisk.
To be able to use this port, the gateway ask for prefix -lets say 9-
then send dial tone and here the user enter the calling number. We want
to cancel this step for the users so they can enter the entire number
In article [EMAIL PROTECTED],
Klaverstyn, David C [EMAIL PROTECTED] wrote:
Thanks for your help.
This is my file.
[channels]
language=au
context=from-pstn
signalling=fxo_ks
This should be: signalling=fxs_ks
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] -
Administrator TOOTAI a écrit :
[...]
FYI, dialing Dial(SIP/exten,,D(0)) give the dial tone, let the user
enter the calling number and the call is passing smoothly.
Sorry, please read Dial(SIP/exten,,D(9))
--
Daniel
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Am Montag, den 11.12.2006, 11:29 +0100 schrieb Administrator TOOTAI:
Administrator TOOTAI a écrit :
[...]
FYI, dialing Dial(SIP/exten,,D(0)) give the dial tone, let the user
enter the calling number and the call is passing smoothly.
Sorry, please read Dial(SIP/exten,,D(9))
Just an
Thanks. What kernels do you use for dom0 and the domU's? Custom-built or
out of the box?
- Arik
jason wrote:
I would vote RAM. I've been using a FXO card in xen for a good year now
with no issues at all. In fact, my zttest timings are the same between
xen and native.
Arik Raffael Funke
Anselm Martin Hoffmeister a écrit :
Am Montag, den 11.12.2006, 11:29 +0100 schrieb Administrator TOOTAI:
Administrator TOOTAI a écrit :
[...]
FYI, dialing Dial(SIP/exten,,D(0)) give the dial tone, let the user
enter the calling number and the call is passing smoothly.
Sorry,
Hi
When trying to install asterisk1.4-beta3 I get the following error when running
./configure:
Cannot find ptlib-config - please install and try again
What is this ptlib-config? Can't seem to find it on google. Where can I find it
and how can I install it? Moreover do I really need it, can I
Network Configurations
Block D, Surrey Park, Barham Road, Westville, 3610
Helpdesk: (086) 163-8266
Tel: (031) 266-1563
Fax: (031) 266-4206
Hi.
Is it possible to have asterisk insert various audio files into the
playback with the music on hold if they are holding on for an extension
or in a
Hi
i have a asterisk server with a Digium 4xE1 card connected to my local
operator.
I am search a How to for :
- Add a Mail to Fax server
- Add a Fax to Mail Server
thanks bye
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ZZ == Zeeshan Zakaria [EMAIL PROTECTED] writes:
ZZ Switches should be Layer 2 or Layer 3, and what's the difference.
You really should hire someone to do the design.
ZZ Another question I have is about 10/100/1000 Mbps. In a standard
ZZ switch, ports don't actually work at 100 Mbps.
They
I have a TDM-400 from digium with 2FXO+2FXS ports. Any idea on how much power
will this drain from the 12 and 5 V connector when all ports are in use?
--
Gustavo Felisberto
(HumpBack)
Web: http://dev.gentoo.org/~humpback
Blog: http://blog.felisberto.net/
It's most certainly
Noc Phibee wrote:
Hi
i have a asterisk server with a Digium 4xE1 card connected to my local
operator.
I am search a How to for :
- Add a Mail to Fax server
- Add a Fax to Mail Server
http://iaxmode.sourceforge.net
http://hylafax.sourceforge.net
Doug
--
Ben Franklin quote:
Noc Phibee wrote:
I am search a How to for :
- Add a Mail to Fax server
- Add a Fax to Mail Server
Oooops, that should have been http://iaxmodem.sourceforge.net
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
Hi folks,
I had a survey online but there i couldnt find a clean sample of CAS
signalling on E1 interfaces. I defined a span with CAS framing and HDB3 line
coding but dont know which signalling to use for channels. I'd use 3 bit CAS
signalling and 20 incoming channels and 10 outgoing ones.
The Digium TE410P base card does indeed work in PCI-X slots. We're
using two of the TE412P's in a PCI-X server with no problems :)
On Sun, 2006-12-10 at 00:10 -0500, Time Bandit wrote:
I can't risk spending a few thousand just to reach the
conclusion that Digium's PRI or BRI cards do not
Hi,
in addition to my previous post about the OSP support on Asterisk,
does anyone know if there existst OSP peering VOIP hosts who are
willing to connect to simple users like me using OSP protocol
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What's the price for these HP switches?
And also I someone can give me a link to some document where I can read
about Layer 2 and Layer 3, how they help in VoIP traffic, it'll be helpful.
___
--Bandwidth and Colocation provided by Easynews.com --
Zeeshan - understanding the Cisco OSI model will help you
conceptualize.http://www.cisco.com/univercd/cc/td/doc/cisintwk/ito_d
oc/introint.htm
There is a good graphic depiction here
http://www.certificationzone.com/cisco/images/graphics/VP/IPTT/WP1/VP-IP
TT-WP1-01.gif Using this image,
On Mon, Dec 11, 2006 at 09:53:53AM -0500, Zeeshan Zakaria wrote:
What's the price for these HP switches?
And also I someone can give me a link to some document where I can read
about Layer 2 and Layer 3, how they help in VoIP traffic, it'll be helpful.
The guy in the UK who bought on Ebay is threatening to buy 2 units
Cory Andrews
Executive Vice President
++
VoIPSupply.com
PBXSelect.com
++
454 Sonwil Drive
Buffalo, NY 14225
voice direct - 716.250.3402
fax - 716.630.1548
e - [EMAIL PROTECTED]
m -
Edgewater Networks markets a 24 port switch, with PoE (both Cisco CDP
and 802.3af supported), and Layer 2/3 management features that retails
for less than $1500. The model is EC-2402POE-01
Cory Andrews
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Check out www.generationd.com for a couple of useful scripts (fax2mail and
mail2fax). If I interpret your question properly, you looking for scripts.
If in fact you are looking for sendmail/libtiff help, have a search through
the archives.
MD
-Original Message-
From: [EMAIL PROTECTED]
Cory Andrews would like to recall the message, [asterisk-users] Re:
Recommendations for QoS, PoE Switches.
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To UNSUBSCRIBE or update options visit:
Gustavo,
Take a look at this thread
http://lists.digium.com/pipermail/asterisk-users/2006-October/169627.html
Presumably the supplemental 12v supply is for ringing voltage.
I did not see anything on Digium's support pages about the card itself.
Maybe a call to tech support may help.
Bob...
On Sun, 10 Dec 2006 20:54:10 -0500
Paul [EMAIL PROTECTED] wrote:
If you run etch before it is released as stable, you might run into
problems that are over your head. I have run into a few that weren't
over my head but they were very inconvenient.
Yes Paul, I'm running 2 etch with asterisk,
Hi all! Do anybody knows any asterisk-dialplan function that can replace
the username portion of FROM header on an INVITE SIP message that is
being handled by asterisk?
Thanks in advance for any tiny clue.
Rgds, Ricardo Martins.
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Does anyone know if asterisk supports PLAR (Private Line Auto Ringdown).
The Oreilly (Asterisk: Future of Telephony) book mentions it in passing
saying that all you need to enable it is to set immediate=yes in
zapata.conf. Has anyone implemented this in brokerage trading
environments?
Thanks
What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter,
the last command in the history always defaults to 'stop now'. This is very
bad, and it's caused accidental shutdowns more than once.
Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399)
Verbosity is
1.I do not have access to console because my servers are in collocation
space, but technician from collocation told me that he is seeing E711 PCI
ERR Slot #1 which in the PowerEdge 1950 manual means The system BIOS has
reported PCI system error on a component that resides in specified slot.
2. I
On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote:
What's wrong with the Asterisk CLI history? When I exit the CLI, and
re-enter, the last command in the history always defaults to 'stop now'. This
is very bad, and it's caused accidental shutdowns more than once.
Nothing wrong here.
On Sunday 10 December 2006 11:18 pm, Remco Barendse wrote:
Hi list!
I have to do a new bare metal installation of a box running Asterisk with
bristuff or vzaphfc.
The box will be used as a really lightly loaded file server and pbx.
Any advise on which architecture I should use? The cpu is
http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l
=enoc=pct3448poe-sapps=bsd
Dell make a nice Poe switch. I've got 20 some odd Cisco 7940G's running
on it at the moment.
-
William J McCloskey
Information Technology Manager
[EMAIL
I use a a400p(tdm400p clone) on a soekris, 2 fxs and 2 fxo, some
soldering needed but the Soekris power supply is enough.
Only the fxs need power, Fxo doesn't.
18v 800 ma
hope it could help
Olivier
Bob Chiodini a crit:
Gustavo,
Take a look at this thread
Strange idea to switch from freebsd to another OS, Freebsd is very
stable with asterisk, I must say, rock solid...
What's the reason?
Olivier
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To UNSUBSCRIBE or
-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]
Sent: Monday, December 11, 2006 10:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CLI History
On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote:
What's wrong
I've also had these problems. If the call is going between two Asterisk
servers, connect them with dtmf=info. That solved my problems.
bp
On 12/10/06, Forrest Beck [EMAIL PROTECTED] wrote:
I too have seen this. I have to press the digits just right. I have
tried RFC2833, and Inband to send
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: Monday, December 11, 2006 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] CLI History
-Original
nobody knows, how jitterbuffer actually working when asterisk doing
protocol translation? i.e. sip-iax, skinny-iax...
how current two jb implementations (generic rtp iax jb) working together?
PJ
Pavel Jezek wrote:
so that, jitterbuffer should be enabled forced on sip and iax
channel on
Jeronimo Romero wrote:
Does anyone know if asterisk supports PLAR (Private Line Auto Ringdown).
The Oreilly (Asterisk: Future of Telephony) book mentions it in passing
saying that all you need to enable it is to set immediate=yes in
zapata.conf. Has anyone implemented this in brokerage trading
On Monday 11 December 2006 9:31 am, Douglas Garstang wrote:
What's wrong with the Asterisk CLI history? When I exit the CLI, and
re-enter, the last command in the history always defaults to 'stop now'.
This is very bad, and it's caused accidental shutdowns more than once.
Connected to
That has been fixed in the current Xen, and as far as I can tell works
without problems. (At least for some NICs I had dedicated to another domU.)
Regards,
Arik
Howard Lowndes wrote:
I have to run Asterisk on the dom0 host as earlier versions of Xen had
problems handing PCI control over to a
It can be configured and DOES work with ZAP channels.
If you are looking to use IP based devices your Mileage may vary from
Hybrid to Sherman Tank.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo
Romero
Sent: Monday, December 11,
Just wondering if there is much CPU overhead in the translation from
IAX2 to SIP, and how taxing this function is as compared to
transcoding.
We're trying to build an efficient system and would like to avoid
taxing the CPU as much as possible. Our upstream service provider is
100% SIP, however
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Carla Schroder
Sent: Monday, December 11, 2006 2:17 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CLI History
On Monday 11 December 2006 9:31 am, Douglas
On Mon, 2006-12-11 at 11:17 -0800, Carla Schroder wrote:
On Monday 11 December 2006 9:31 am, Douglas Garstang wrote:
What's wrong with the Asterisk CLI history? When I exit the CLI, and
re-enter, the last command in the history always defaults to 'stop now'.
This is very bad, and it's
The protocol does not matter. If jitterbuffer is off then asterisk gets the
packets and sends them to the IAX clients without jitterbuffer just as if it
was another SIP client w/o jb.
On 12/8/06, Pavel Jezek [EMAIL PROTECTED] wrote:
so that, jitterbuffer should be enabled forced on sip and
Hello all,
I'm having a bit for a problem with the dial command limit option. I have
the following dial command (executed from inside the a2billing agi)
AGI Script Executing Application: (Dial) Options: (
IAX2/[EMAIL PROTECTED]/18005551212|30|HL(6:2:0)0)
Now, from what i read in the
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Monday, December 11, 2006 12:57 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CLI History
On Mon, 2006-12-11 at 11:17 -0800, Carla Schroder wrote:
On Monday 11 December 2006 9:31 am,
- Original Message -
On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote:
What's wrong with the Asterisk CLI history? When I exit the
CLI, and re-enter, the last command in the history always
defaults to 'stop now'. This is very bad, and it's caused
accidental shutdowns more
Dear Friends and Supporters!
I am trying to upgrade my testing asterisk realtime 1.2.13 with MySQL 5.0 and
unixODBC to the beta asterisk 1.4.
I run the make and make install for the asterisk-addon just fine, It created
the modules res_config_mysql.so and cdr_addon_mysql.so without any problem
this problem is being actively worked on right now in mantis(bugs.digium.com),
your best bet is to monitor the issue while it's being worked on. and test the
any patches as they are uploaded
-anthony
- Original Message -
From: Yair Hakak [EMAIL PROTECTED]
To: Asterisk Users List
short version: me too
long version: The same thing happens on my asterisk boxes - both
built with the latest trixbox image... perhaps that's a factor? My
history is always restart now, although I typically connect and run
sip show peers. I haven't typed restart now in a long time, but
-Original Message-
From: Mailing List [mailto:[EMAIL PROTECTED]
Sent: Monday, December 11, 2006 1:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CLI History
- Original Message -
On Mon, 2006-12-11 at 10:31 -0700,
hello there,
I wonder if you were able to over come your problem in configuring your aculab
card?
Ammar Ali
From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Thu, 20
Apr 2006 16:44:38 +0100 Subject: [Asterisk-Users] zaptel and zapata
configuration Hi I am trying to
Hi All,
Has anyone hooked up * as an extension/trunk of an Avaya system that has
around 2 ISDN30e's.
Trying to add 100 extensions to one of our systems, but not sure where to
start reading.
Thanks.
--
Kind Regards,
Gavin Henry.
___
--Bandwidth and
Hello Trevor,
I wonder how I can find out for sure what is the H/W version for a PROSODY
ACULAB SS7 Card? I dought that I have a ver 1.1 which have may issues with
recently made computers.
I have a case opened for my problem with aculab but sysdiag shows that I have
ver 1.1 and aculab
Don't hit Ctrl-C!
If I type ? in the CLI, Ctrl-C is not listed as a command.
*CLI
! abort add ael agent agi cdr
databasedebug dnsmgr dontdumpdundi
extensions
feature group helpiax2include
Any idea what causes the warning Unable to open pseudo channel for
timing... Sound may be choppy.? Any ideas what I need to resolve
this? I do have the zaptel module installed but don't have a zaptel
card. I'm guessing this has to do with ztdummy? I'm running Debian and
installed asterisk,
Im passing a PVR-500, a PVR-250, a dual Intel Pro100 NIC (2 interfaces)
one of the onboard IDE controllers, all of my USB ports and my FXO card
without any hiccup. I stay pretty bleeding edge, so I can't say if this
would work out of the box. I did have to tweak a few PCI latency timers
but
But ctrl-c is 3 less keystrokes than exit\n !
-Original Message-
From: Steven [mailto:[EMAIL PROTECTED]
Sent: Monday, December 11, 2006 2:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: CLI History
Don't hit Ctrl-C!
If I type ? in the CLI, Ctrl-C is
here is the latest update:
in zaptel.conf i used
fxsks=1-4
fxsks=5-8
fxsks=9-12
fxsks=13-16
zttool shows hardware OK
ztcfg worked normally
in zapata.conf when i define the channels channel=1-16 and restaring
asterisk it gives the below errors:
Dec 12 00:48:28 WARNING[3141]: chan_zap.c:921
I figured out the problem, it is the location of FXO boards on cards,
channels are from 9-24 not 1-16.
Thanks all for your help, specially Tzafrir, genzaptelconf shows it clearly.
On 12/11/06, O. Kamal [EMAIL PROTECTED] wrote:
here is the latest update:
in zaptel.conf i used
fxsks=1-4
DG == Douglas Garstang [EMAIL PROTECTED] writes:
DG When I exited the CLI and re-entered and pressed ctrl-c,
That's where your problem is. Use exit and not ctrl-c to leave
asterisk -r.
/Benny
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In chan_sip.c, line 5876 (Asterisk-1.2.13), the
function parse_ok_contact returns whether the host
that requested an invite is a valid or invalid host.
In line 5925 the following clause is tested:
if (!(ast_test_flag(pvt, SIP_NAT) SIP_NAT_ROUTE))
hp = ast_gethostbyname(n, ahp);
If this
Hi All
Could a VPN be used to help with SIP Tunneling and QoS issues.
State 1:
Two IP Networks Connected via the Public Internet transmitting VoIP Traffic
Say a VoIP User and VoIP Termination Provider.
Each side can put QoS onto their part, but if QoS does NOT exist between
them
then call
Hi David
Care to share how you approached using Diffserv and VLANs with the FSM7326P
We are considering the same switch. But I'm unsure about the configurations
required.
Thanks in advance
Barry
David Coulson wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Typically we deploy the
It looks to me that if the test clause is false then
ast_gethostbyname is called. Presumably not needed when NAT is enabled.
Bob...
je . wrote:
In chan_sip.c, line 5876 (Asterisk-1.2.13), the
function parse_ok_contact returns whether the host
that requested an invite is a valid or
I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my
polycom phones and then it also sends 183-Session Progress. That doesn't seem
to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ?
Doug.
___
Hi Barry,
I used SIP over OpenVPN when travelling, especially from hotel rooms or
showfloors. Of course I did not expect the performance of a local SIP
connection, but generally it worked OK. The latency would not suffer
much in comparison to direct connection, but a WLAN was involved which
would
When we send 183, that means 'inband progress' is available. That does _not_
necessarily mean that it is ringing, it could be any sort of progress tone,
or even audio from an IVR. If your ATA does not stop its own ringing
generator and start forwarding the audio, it is broken.
It is my
So in your example you can manage QoS within the VPN but have no control
whatsoever over the VPN tunnel as a hole, it would be the same result as if
you just passed straigth TCP over your connection with QoS, however you
will waste more resourses for the VPN and probably introduce a bit of
You need to understand how NAT works, if you can chan2 and chan2 is behind a
NAT and suddenly someone else is invited to chan2's IP address port 5060
chan2's router willl say WTF I dont have an estabished connection on port
5060 (to the client being reinvited to chan2) and it wont work. You need
Hi Anselm
Thanks for your input
Yes I was thinking of using OpenVPN so it was good to hear your experiences
I'm not so much concerned with the encryption of traffic etc..
But the Level of QoS.
If my IP Phone set QoS and the VoIP Termination provider's * PBX sets QoS
And we now connected via a
Some VPN implementations allow you to copy the ToS of the encapsulated
packets to the ToS of the wrapper packet.
Andrew Joakimsen wrote:
So in your example you can manage QoS within the VPN but have no control
whatsoever over the VPN tunnel as a hole, it would be the same result as
if you
Hi List:
I can not find out an example how to store include = context name
statement into Realtime static.
Please help me on this one.
Thanks,
Tielin
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asterisk-users mailing list
To
Andrew,
I don't think it's a Polycom issue. We took Asterisk out of the picture and had
our Polycom phones communicate directly with an Audiocodes PSTN gateway. Unlike
Asterisk, the audiocodes do not send 180 Ringing before sending 183 Session
Progress, and the polycom's play the correct
Douglas Garstang wrote:
Andrew,
I don't think it's a Polycom issue. We took Asterisk out of the picture
and had our Polycom phones communicate directly with an Audiocodes PSTN
gateway. Unlike Asterisk, the audiocodes do not send 180 Ringing before
sending 183 Session Progress, and the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mochamad Susantok wrote:
Hi all,
Howto configure asterisk 1.2.13 (debian-base) with support Instant
Messaging, especially using client Xlite v.3.
Thanks
Hello,
Im using my patched chan_sip.c for that.
My mistake, I misread it. So if a hostname is provided
(e.g. [EMAIL PROTECTED]) instead of an IP (e.g.
123.123.123.123) and the recipient of the INVITE is
not using NAT then ast_gethostbyname will be run - is
that correct? In this case, why the distinction
between a NATted and non_NATted
[channels]
context=default
signalling=fxs_ls
;channel=1-16
usecallerid=yes
hidecallerid=no
callwaiting=yes
restrictcid=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
;accountcode=lss0101
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
To the best
-Original Message-
From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED]
Sent: Monday, December 11, 2006 5:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Sends 180-RINGING to UAeven
withprogressinband=yes
Douglas
Douglas Garstang wrote:
No... but if we answer the call before dialling, isn't that going to cause a
whole world of billing hurt?
You are only answering the call leg from the Polycom to Asterisk. You
are not answering the Asterisk - PSTN leg (I assume that is the only
leg you bill for)
On 11 Dec 2006, at 04:25, cb wrote:
I recently purchased a Mediatrix 1124 from an auction of a company
that went out of business. It came with nothing other than the unit
itself.
In digging thru the Mediatrix web site, and various google
searches, it looks like it only supports SNMP
Thanks for everybody's help.
Cory, thanks for the links. I once studied OSI model, many years ago, when I
was doing MCSE for Win NT. I'll go through these Cisco documents to
improve/update my knowledge about OSI layers and see how it can help me in
VoIP networking.
Reorder tone can be used for many things, is there anything I've missed?
7.4.2 401 Unauthorized 78
7.4.4 403 Forbidden ... 78
7.4.5 404 Not Found ... 78
On Dec 11, 2006, at 8:58 PM, Tim Panton wrote:
It looks like there might be enough info on these pages to get you
going:
Thanks for the links! Hopefully I can get somewhere with the info.
If you need a hand with the SNMP side, drop me a mail
I'm pretty new to SNMP, so I may take you
How do i patch file chan_sip.so ?
I use asterisk with Debian distro not asterisk-XXX.tar.gz
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Mochamad Susantok wrote:
Hi all,
Howto configure asterisk 1.2.13 (debian-base) with support Instant
Messaging, especially using client Xlite v.3.
If my IP Phone set QoS and the VoIP Termination provider's
* PBX sets QoS. And we now connected via a VPN tunnel.
We should be able to guarantee Quality due to the Tunnel.
Nope. You only control the QOS within your tunnel (i.e. among other
traffic flowing through the tunnel). But what QOS
Just 'sox -v 1.5 beep.gsm loudbeep.gsm' ?
CP
On 2-Dec-06, at 11:29 AM, Peder @ NetworkOblivion wrote:
We've had a few people complain that the beep before leaving a
voicemail is not loud enough and too short. Does anybody have a
recorded beep that they can share, that is a little louder and
Hmmm. Ok, that's true. At the very least it will create confusing CDR's I
think... maybe. We're not billing our OnNet traffic at all. Only the traffic
that goes OffNet, to our switch is billed (if it leaves our switch that is...).
I was thinking earlier too that we only need progressinband on
You can run Asterisk 1.2 in sarge using the packages in backports.
Just add:
deb http://www.backports.org/debian/ sarge-backports main contrib non-free
to /etc/apt/sources.list
then apt-get update
and then apt-get -t sarge-backports install asterisk
(you can also pin-priority asterisk's
On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote:
What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter,
the last command in the history always defaults to 'stop now'. This is very
bad, and it's caused accidental shutdowns more than once.
thats prety
Hey everyone !
I have a problem in making outbound calls in PRI connection.
I have E1 PRI airtel connection [ India ]
[ asterisk-1.2.12.1 on CentOS 4.4 ]
zaptel.conf
--
[channels]
language=en
usecallerid = yes
hidecallerid = no
callwaiting=yes
threewaycalling = yes
usecallingpres=yes
Hi all,
Thanks for your reply,
I'm using sip communicator(in java that is intergrated with.. ) and asterisk
is interfaced with this.
i'm able to make calls between pingtel and Voip user,
and also i can able to make call from Sip communicator to pingtel or Voip
phone.
but now i'm can't make calls
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