[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
I will publish a tutorial in the beginning of next week about how to
configure Zoiper and Asterisk to do t.38 together.
Zoa.
Where will you publish this tuto ?
Regards
- the CDR's aren't reliable enough for a billing
platform (as you've rightly pointed out) but are OK for very basic call
logging (something the customer can look at).
Hopefully, the murf'ster will chirp in here :).
Cheers
Andy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
support, please also include sip and udtpl captures.
Cheers,
Zoa.
Olivier wrote:
2008/12/5 Stefan Lekov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Olivier wrote:
2008/12/3 Olivier [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
mailto:[EMAIL
Odd problem, where some remote phones, at users homes, dial and connect fine,
no matter what the destination is.
Bad phones, SIP to SIP, between remote and office, or remote to remote, work
and have good audio, but no audio, at all, to PSTN or Cell phones.Phone can
be moved to office and
Billing and logging should not be confused theoretically - I agree. But
in practice,
the logging of the calls (not other events of the system) IS used for
billing purposes.
The start and finish time is not enough for many (I not that it is not
enough for me).
The accountcode is not enough for
I tried sending faxes through Zoiper (Zoiper to Zoiper) last week
and the program crashed. After an update it stopped crashing but
still could not send a FAX. I then tried Kapanga (the free version has a
limited 30
days FAX sending capability) and it worked. This might be of little use
to you
I would go for chan_h323. Much more stable since 1.4
and the config more close to the other channel configs too.
We used it on production for a long time and it worked well
although a little heavy cpu-wise. To get started you need to install
openh323
and pwlib from here
I agree with Freddi and would like to add that a field indicating the
order of the outgoing legs would be very useful. For billing purposes
one could benefit very much if one new the order of the providers
that were called in a specific call.
Freddi Hansen wrote:
To me the obvious answer is to
that are
associated with that call. Your select grabs the first dimension, a list
of all calls. Then using the unique identifier of each call you build a
second dimension of the related calls.
[EMAIL PROTECTED] wrote:
In order to avoid a multidimensional schema we could have 1 cdr per call
leg. So
I think that the custom cdr back-end can be successfully used to
maximize or minimize the CDRs detailing
on a per-needs basis. Furthermore, the CDR() function gives plenty of
room for even more detailing.
In my opinion the detail level (fields) is not the issue with the CDRs
generation nor is
. And of course a keep it simple stupid mode that just writes the
top level cdr as it does now.
[EMAIL PROTECTED] wrote:
I think that the custom cdr back-end can be successfully used to
maximize or minimize the CDRs detailing
on a per-needs basis. Furthermore, the CDR() function gives plenty of
room
If we only implement A-D cdr we lose information.
On the other hand, if we implement all 3 CDRs for one call we can
either use this info or ignore it and act like its not there. The first way
is prohibiting for some users. The second one can match any scenario
with none to little effort.
Steve
On 11/22/2008 at 11:17 PM, Barry L. Kline [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have an Asterisk box sitting between the PSTN and a legacy PBX. I
have successfully configured Asterisk to use IMAP for voicemail and have
written the necessary script
Hello List,
i would like to set up the following concept:
Scenario 1:
=
VOIP-Phone -tcp/udp- VOIP-Phone
(direct P2P between two phones. Those phones have be he hard phones.
No Software such as KPhone or something)
Scenario 2:
=
VOIP-Phone -tcp/udp-
Hi Valetin,
Valentin Bud wrote:
Are the VoIP phone mobile on the internet or in fixed locations?
If they are in fixed locations and they have to go through internet to reach
the asterisk box, the way *i* would do it is with VPN tunnels. If they
are in the same
location (LAN) it is very
Hello,
I'm the person responsable for the zoiper roadmap, comments inline
snip
This all started because Zoiper really annoyed me - they keep sending me
beta versions of their software (which is nice, thanks you), and they keep
on compiling it for ubuntu or some other distribution of linux
Hi,
I want to be able to bridge two sip channels using direct RTP
between my endpoints (Audio IP : not local) but without
using reinvites. So I set up my asterisk sip endpoints as follows:
[test1]
type=friend
host=dynamic
username=test1
dtmfmode=info
context=test_rtp
allow=all
Yes, but my conf is quite straightforward, isn't it?
No NAT etc...
I just want to know what is the combination of directives that I have to use
in order to achieve my goal.
Is there going to be any support in the future for this feature?
Because from the little I' ve seen in the mailing lists
I'm pretty sure it is a VERY OLD ISDN router and you can't use it with
*. The voice ports have no VoIP capabilities, they are just used
directly from the ISDN line.
Ronny Julian wrote:
I found this at a local sale. I need to find a power supply for it.
Before I do I wonder if anyone can
Hello all,
I would like to use say.conf settings but every time i restart
asterisk i have to load manualy say load new is there a way to do it
automaticaly i use asterisk 1.4.19
Thanks
___
-- Bandwidth and Colocation Provided by
You generally don't need to enter the public IP of the router into the
Cisco, just setting nat_enable to 1 is almost always sufficient. * is
smart enough to realize that the IP of the packet is the public IP of
the phone.
Tony Mountifield wrote:
I have used Grandstream phones for years, and
I am running 1.4.10.1 and I am getting garbled MOH from calls within the
same LAN with no firewall. Calls sound fine, but every 5-10 seconds the
MOH gets garbled. I am using the stock MOH files. Any ideas where/how
this could occur? There is no debug showing any issue with MOH. Thanks.
.
Matt Gibson wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Wednesday, October 08, 2008 10:13 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] cisco phones getting SIP 401 unauthorized
Hi Jerry,
Hm
As a followup to my previous email, change nat_enable to 1 and reboot
the phones.
Jerry Geis wrote:
Did you check sip.conf to make sure that the port is correctly set to 5060?
Please show the output of Cli sip show peer peernumber and the contents
of your SEPMAC.cnf file.
Dave
Kevin P. Fleming wrote:
Olivier wrote:
2. R Hook-flash key is now available to transfer calls.
In s450IP web management server, its defaults settings are :
Application-type: dtmf-relay
Application-signal: 16
Is there anything to configure in features.conf, extensionsconf or
elsewhere
They are probably referring to the fact that the base 7960 is End of
Life and the 7960G is probably going to be EOL soon as well, so they
won't offer new firmware at the EOL milestone. They have been replaced
by the 7961. Completely different firmware and configuration, but there
still is
I've had the same experience. I probably have 20-30 customers with
multiple SIP phones behind PIX running 6.3(5) (which has been out almost
3 years) and I have no issues at all. You can even have two phones
behind a PIX being PAT'd to a single external IP with reinvite enabled
in * and you
I've got a problem that I hope someone here can shed some light on.
It seems that in any calls going over Zap channels (either with a FXO card or
PRI card),
inbound audio is constantly monitored for DTMF tones, and then these tones are
regenerated back in the audio stream either within * if
Hi,
I decided to migrate my scripts from DeadAGI to AGI (FastAGI).
The no-exit-on-hangup behavior suited me just fine with DeadAGI.
How can I make my AGI scripts (which are executed on another AGI server)
NOT to exit when a hangup is detected?
I used AGISIGHUP=no before calling the AGI
Hi All,
would Asterisk 'transcode' H.245 alphanumeric DTMFs
to an H.245 signal / rfc2833 H.323 device over G.729 codec ?
Thanks for supporting,
.TF
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22
A couple of years ago I started my Asterisk carrier with selling
x100p cards and I think I sold around 100 of them in total to people
who could actually contact me and new who I was. Yes, it is a poor
man solution but at least it is a solution. And for the poor man it
is the only thing
An option to rotate between numbers is to add a queue to the system
and add and as agents and pick the proper strategy (rrmemory
or leastrecent). This has some advantages:
- the calls are devided as you have in mind
- when there are more calls coming in they are queued instead of
) rings 10 seconden
exten = 2,n,Playback(plaza/external_transfer) ;
exten = 2,n,Dial(IAX2/[EMAIL PROTECTED]/0621831234,10,t) ; using
iax2 trunk OOO50608 external number 0621831234 is called for 10 seconds
exten =
2
,n
,VoiceMail(5001
Dinesh Nair пишет:
On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:
The best option is to put a SIP Proxy in front of your Asterisk sever
and block REFER requests.
or just comment out the block in chan_sip.c which handles the refers.
Thanks to your answers, but i found
Hi folks,
I have some asterisk 1.2 box with self-made billing, and I need to
disable call transfer on all calls and directions.
I turned it off in features.conf and there is no 'tT' option in all my
Dial() commands, but users still able to transfer call using transfer
function in ip of
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
]On Behalf Of Gonzalo Servat
Sent: Friday, March 21, 2008 12:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] www.cdsportal.net wholesale
voipprovider --starting
I have absolutely no idea since I was not even aware of it. However,
this may give you some hints as to where you can find more information:
http://www.mail-archive.com/[EMAIL PROTECTED]/msg27124.html
- Waldo
On Feb 22, 2008, at 5:08 PM, Douglas Garstang wrote:
It's time to ask
that this is
happening. This problem started when I added the groetstraat configuration.
TIA,
Hans Feringa
zuiderven asterisk = 1.4.18 (compiled from source)
groetstraat asterisk = 1.4.10 (ubuntu repository)
This is the local (zuiderven) iax.conf:
register = **:[EMAIL PROTECTED]
register = 8*:[EMAIL
I would like to know as well, it has never worked for me.
On Dec 18, 2007 4:27 PM, JR Richardson [EMAIL PROTECTED] wrote:
Hi All,
Anyone know the sip header to send to a Linksys to resync it's config
file?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
You can also do
asterisk -rx stop gracefully
From any sort of script / crontab, etc.
On Dec 10, 2007 10:36 AM, Jeng Yu [EMAIL PROTECTED] wrote:
Thanks, All! And thanks, Oquendo! I will experiment
with this suggestion. I was actually thinking in terms
of a situation where it would be done
of not
requiring any changes to your asterisk configuration.
On Dec 7, 2007 9:12 PM, Philip Prindeville
[EMAIL PROTECTED] wrote:
That's sort of my point: that you have to reinvent it, and it's easy to
get wrong.
___
--Bandwidth and Colocation
On Dec 7, 2007 11:28 AM, Michael Munger [EMAIL PROTECTED] wrote:
Is there anyone interested in developing an open source Asterisk / MS
Exchange solution?
Please explain. This sounds interesting. But why MS exchange only? I
think it's safe to say with good IMAP and LDAP support we can
integrate
options any application that uses TAPI for placing
calls will work with no added configuration/modification.
There are various 3rd party TAPI drivers but I think these little
things are items that need to be added to the Asterisk development.
On Dec 5, 2007 12:26 PM, Jared Smith [EMAIL PROTECTED
What don't you tell us what you are ultimately trying to do. You want
the callerid next to the connect message in debug output... why? What
will that help you to accomplish?
On Dec 7, 2007 4:42 PM, Arpit Mehta [EMAIL PROTECTED] wrote:
Ok so the call reference is the 'cr' field (q931.c) and how
OkWhat is the issue? Does your G729 not work?
Anyways who cares about the CPU? If you have a 32 bit Linux you need a
32 bit program.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE
Besides grandstream-doorphone transplant surgery, no. But it does have
PoE. It's cheap, especially if you already have a doorphone. If you
used a GXP-2000 you can use the display and it supports XML idle
screens.
On Dec 4, 2007 2:53 AM, Nick Seraphin [EMAIL PROTECTED] wrote:
On a similar note
Griefs?
rejected connect attempt from 111.111.111.111, who was trying to reach
'12345678' No authority found
call rejected by 111.111.111.111: No authority found
But once it works it works...
I have DTMF issues with sending calls from 1.2 to what I suspect is a
really old 1.4 build via IAX that
If the majority of the MoH is queues, move the location of the queue.
On Nov 28, 2007 4:42 PM, Darryl Dunkin [EMAIL PROTECTED] wrote:
Does anyone have any opinions on the music on hold quality over G729?
The stock files seem to sound terrible over it, this is enhanced further
by calls coming
Anyone have an idea how to implement a phone number that can only be
called once? The first time it will process normally and any
subsequent calls will be rejected.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
[Nov 28 15:42:41] WARNING[4098]: chan_sip.c:4957 process_sdp: Unable
to lookup host in c= line, 'IN IP4 50045'
Anyone have this problem when using T.38 faxing... and some solution perhaps?
___
--Bandwidth and Colocation Provided by
I have installed an Asterisk 1.4 on Suse93 using a FritzCard.
Some calls are logged to the ISDN log, but Asterisk is not detecting
incoming calls.
I wonder whether some other device or process is preventing Asterisk
from gaining access to the isdn line?
Is there some way to ensure that only
Nortel config.
Jonn
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Wednesday, November 28, 2007 2:06 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk - Nortel Phone Switch
Still trying to make my Asterisk
(telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.10:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK489024dd;rport
From: Shawn Ip sip:[EMAIL PROTECTED];tag=as25dd7670
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL
On Nov 26, 2007 9:52 AM, Alberto Pastore [EMAIL PROTECTED] wrote:
I also found the Pirelli DP-L10 dual phone to be an excellent sip client
with good roaming support and discrete battery saving capability.
(Used in a 14-cell wifi network with 40 cellphones).
I don't know what to say I have
Your form can no longer accept submissions.
SuSe 10.1 with latest Asterisk 1.2 using our own patches.
We are about ready to go live with new installations of SLES or CentOS
+ Asterisk 1.4 just need to work out the bugs.
On Nov 26, 2007 5:14 AM, randulo [EMAIL PROTECTED] wrote:
Hi,
I'd like
Hi all,
Im preparing a quote for a 5 Star hotel, planning to have around 100
SIP Wifi phones for PBX operations running on 100 AccessPoints.
Network is running in ARUBA Networks - AP70 access points.
The initial recommendation is to go for Hitachi Wifiphones, but i
would like to know from the
Hi all,
Im preparing a quote for a 5 Star hotel, planning to have around 100
SIP Wifi phones for PBX operations running on 100 AccessPoints.
Network is running in ARUBA Networks - AP70 access points.
The initial recommendation is to go for Hitachi Wifiphones, but i
would like to know from the
vary I don't believe that forum posting
was made in any official capacity but I also doubt that Digium would
not do something to correct an issue for an item under warranty.
On Nov 22, 2007 8:03 AM, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
Is Digium the best telephony cards to be used
Dear Bruno,
I had the experience of using the Vcidial with the boards of Digivoice.
It worked very well!
Leonardo Silva
Does Vicidial work together with Unicall/mfcr2 ?
Best Regards
--
Bruno de Assumpção Loureiro
msn: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED
I just bought an Aastra 480i CT for a client who needed cordless
capabilities in their office. I'm trying to set up the base station and
cordless handset in my office first. I'm able to connect the phone to
my Asterisk box and make outgoing calls from either the base station or
the handset -
I figured it out. Unlike the Linksys SPA942, the Web GUI interface for
configuring the phone requires Proxy Server as well as the Registrar
Server fields be populated with the IP address of the Asterisk server.
[EMAIL PROTECTED] wrote:
I just bought an Aastra 480i CT for a client who needed
You need a T38 gateway of sorts, sort of like the app_t38gateway of CallWeaver.
However digium refuses to include such a program with Asterisk.
On Nov 21, 2007 6:13 PM, Robert Moskowitz [EMAIL PROTECTED] wrote:
It seems that Spandsp has everything in it (when you include rxfax and
txfax
Take a look at the admin guides at http://spc.pifiu.com
On Nov 18, 2007 10:53 PM, Philip Prindeville
[EMAIL PROTECTED] wrote:
I'm using a bunch of SPA942's, and I'm trying to provision them mostly
by DHCP (and what I can't set that way, I try to provision via HTTP
interface into the phone
You need some experiance with the ANSI C programming language. Once
you have acquired that the rest is pretty straightforward.
On Nov 14, 2007 2:21 AM, Rilawich Ango [EMAIL PROTECTED] wrote:
You mean modify the source? Could you give me an example, say I wrong
to remove advance option
did you try
canreinvite=no
in your sip.conf file
It would also help to:
1) Post the relevant configuration files (phone AND Asterisk)
2) Post the EXACT message from column 1 to EOL
3) What version of Asterisk? Stock? From a certain distribution? Patches?
Or I could just say There is a problem
vi app_voicemail.c
On Nov 13, 2007 10:34 PM, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi all,
Can I simply the voicemailmain IVR? I just only want some of the
option in voicemailmain, ie read or delete messages. Is it possible
to configure that function?
Ango
complies with the relevant
standards) faxing will work with 100% reliability, thats a bit more
assuring that it should work, no?
On Nov 10, 2007 7:34 AM, Greg Cockburn [EMAIL PROTECTED] wrote:
Hi all,
the company I work for has an aging Digital PBX attached to an E1.
This PBX has a few analogue
Broadcom BCM5721 working here with SuSE (2.6.16)
On Nov 9, 2007 12:39 AM, Michael J. Liberatore
[EMAIL PROTECTED] wrote:
Hi, I am getting a new sangoma t1 card soon and that will max out my slots,
which means i need to take out a card. I am going to take out my pci
network interface card
On Nov 8, 2007 7:11 PM, Philip Prindeville
[EMAIL PROTECTED] wrote:
Anyway, pointers for someone wanting to learn to quickly diagnose SIP
conversations would be great.
Read rfc2543, rfc3261 rfc3265. Otherwise what you want to do is akin
to trying to diagnose a nuclear reactor and not wanting
Setup a 2nd registration on the phone that only allows 1 call at a
time. Ideal setup it up as a shared appearance so call forwarding,
etc dont work on that registration. This way your phone has 2
registrations 1 for any direct call and another for shared calls,
queues, etc.
subscriptions, show
hints, show channeltypes and SIP show inuse.
On Nov 6, 2007 1:36 PM, Alejandro Cabrera Obed [EMAIL PROTECTED] wrote:
Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The
SIP clients are using different operating systems such Debian, Gentoo
and Windows XP so
On Nov 6, 2007 2:25 PM, Jarga Jallow [EMAIL PROTECTED] wrote:
Under asterisk info: Sip registry 12/12 76.xxx.xxx.xxx
D N 5066 UNREACHABLE
11/11 76.xxx.xxx.xxx D N 5064 UNREACHABLE
10/10 76.xxx.xxx.xxx
For fastest handover disable any sort of encryption and use the same
SSID for all AP... infact I don't know how you would setup roaming
otherwise. Channels don't have to be the same, but optimize for the
best RF performance/least channel overlap.
___
Have you looked at your TFTP server logs?
On 11/7/07, Roi Stork [EMAIL PROTECTED] wrote:
I am currently testing a 57i unit. No problems configuring the phone's
config via phone/web UI.
We are trying to avoid using the web UI, the reason is it will take a
long time typing the softkey xml
Have you tried a second unit? I don't trust the Grandstream ATA at
all. We only bought 3 but none worked!
On 11/7/07, Per Jessen [EMAIL PROTECTED] wrote:
I've got a Grandstream 487 in a home-office. The phone-side is working
fine, but the user is complaining that his internet connection keeps
Add qualify=5000 in the relevant section of your sip.conf (under the
[6464]) and also make sure the phone is configured NAT Keep Alive
Enable = YES.
On 11/7/07, Kim Joung-il [EMAIL PROTECTED] wrote:
Sorry guys, I should have already sent such details...so
1. Yes, device is behind NAT
It should be possible to get the video call over PRI or ISDN and
depending on the codec in theory it could just be throwing packets
into SIP.
On 11/1/07, voip Server asterisk [EMAIL PROTECTED] wrote:
Hi..
Iam new with asterisk PBX, and i have read about asterisk video call.: my
question:
1
? Which version of
Asterisk?
On 11/7/07, Doug [EMAIL PROTECTED] wrote:
At 02:47 11/7/2007, CSB wrote:
Content-Type: multipart/alternative;
boundary==_NextPart_000_0007_01C82187.BC96F350
Content-Language: en-nz
How is the billsec field calculated in CDRs?
I have a situation where
On 11/7/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
I'm not really sure if Callweaver has this limitation or not. But they
did aim at using high-resolution timers from the Linux kernel.
Callweaver does. Asterisk does not. I'm awaiting their next release
its supposed to have proper faxing
router.
However I have yet to find a WiFi handset I am happy with.
On 11/6/07, Michael Graves [EMAIL PROTECTED] wrote:
I'd like to survey those on-list who actually use wifi SIP handsets.
What type of wifi access point do you use? Are you happy
Post the relevant configuration files we'd be glad to help.
On 11/6/07, Kim Joung-il [EMAIL PROTECTED] wrote:
Hello!
We are using several Linksys SPA-941 in our office. After IP change occur
devices seems not to be reachable, actually unavailable! Devices is
connected, e.g. we can place
On 11/6/07, Hans Feringa [EMAIL PROTECTED] wrote:
I understood that a timing device (ztdummy if no zaptel hardware is
present) was not necessary anymore with linux kernel 2.6.
When I enable iax2 trunking I get this warning
chan_iax2.c:8908 build_user: Unable to support trunking on user
What's the result if you do cat /dev/zap ?
On 11/6/07, Paulo Garcia [EMAIL PROTECTED] wrote:
Hi,
I have these two cards, the Sangoma has 4 fxo interfaces and the
digium has 1 fxo and 1 fxs.
After install the sangoma card, my zaptel.conf was configured for that
card. I'm trying to configure
Sorry I mean ls /dev/zap
On 11/6/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
What's the result if you do cat /dev/zap ?
On 11/6/07, Paulo Garcia [EMAIL PROTECTED] wrote:
Hi,
I have these two cards, the Sangoma has 4 fxo interfaces and the
digium has 1 fxo and 1 fxs.
After
Just remember if you don't have any Zaptel cards you are going to have
to use ztdummy to run app_meetme. Ztdummy essentially requires Linux
2.6, which you should be using anyways.
On 11/6/07, Carles Pina i Estany [EMAIL PROTECTED] wrote:
Hello,
First of all: also thanks to Doug Lytle
On 8/29/07, Steve Underwood [EMAIL PROTECTED] wrote:
Carlos Chavez wrote:
On Wed, 2007-08-29 at 00:03 +0300, Tzafrir Cohen wrote:
On Tue, Aug 28, 2007 at 10:11:03PM +0200, Christian Peter wrote:
Hi list,
I'm running current SpanDSP
http://www.soft-switch.org/downloads/spandsp
On 10/31/07, Michelle Dupuis [EMAIL PROTECTED] wrote:
The T1 was setup as tie line, not a trunk. The Bell guy tried setting up
the line 2 ways:
1. As a trunk. This did not work because:
a) When he typed in the access code for the trunk on a phone set (and
then any numbers), the call
On 10/31/07, Gordon Henderson [EMAIL PROTECTED] wrote:
Not strictly asterisk related, however...
Here's an odd one for you.. I got a Nokia E90 and setup it's SIP client
which runs via Wi-Fi (anyone know if I can make it work via GPRS/3G?)
Anyway, in a fit of idleness, I thought I'd see what
Steve Underwood wrote:
SpanDSP cannot be used by the standard distribution of Asterisk, as it
is GPL code. However, if you are using Asterisk within the restrictions
of the GPL you can add app_rxfax and app_txfax to Asterisk quite easily.
I was wondering how someone could modify Asterisk
any settings no matter what and not allow
a factory reset... paperweight.
I'd personally use Polycom in the situations where there's no NAT and
the Linksys SPA-phones where you do have NAT.
On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi all,
We have a client that needs to setup
I thought there was some talk of getting T38Gateway into asterisk_addons?
Stupid linking bullshits.
On 10/31/07, Paul Bryson [EMAIL PROTECTED] wrote:
Nasir Iqbal wrote:
Hi,
Have you tried Callweaver http://www.callweaver.org
I was really hoping to be able to use Trixbox to do
Here's a link to the free version:
http://asterisk.hosting.lv/
On 10/31/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Tue, 30 Oct 2007, satish patel wrote:
Dear all
I have already post this question but i need more input for
this setup
[IPphone]--[Asterisk
/25/07, Pepo [EMAIL PROTECTED] wrote:
I am trying to use Asterisk as the voicemail system of the TELCO where I work.
I wanna test with 2 mail boxes ( and later with a better machine/server I
hope try with 7 ).
How do I include in voicemail.conf the file with the mail boxes?, In a big
Hi all,
We have a client that needs to setup about 80 desk phones (about 50
in one location and about another 30 in 5 different locations). Which
brand/model would you recommend. We were personally thinking in
recommending either Cisco, Aastra, Polycom, or Snom, for we've heard
great
My apologies to the list for not having entered a subject line in the
email.
Thanks
On Oct 29, 2007, at 1:42 PM, [EMAIL PROTECTED] wrote:
Hi all,
We have a client that needs to setup about 80 desk phones (about 50
in one location and about another 30 in 5 different locations). Which
Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, October 29, 2007 10:42 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users
No:
register = abc:[EMAIL PROTECTED]
[peer]
host=zzz
Its possible to make mistakes and typos you know. Maybe you can post
your config file and we can help you.
On 10/26/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi Pablo;
How the IP address will be wrong, and asterisk able to
do
Take a look at http://spc.pifiu.com there they have the spc.exe (
Linux variant) which will generate the sample XML file for your
firmware version. There is also in PDF format the admin guides that
explain all the parameters.
On 10/29/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
i need
Or you can download them at http://spc.pifiu.com and not have to go
through that bullshit.
On 10/29/07, John Mason Jr [EMAIL PROTECTED] wrote:
If you go to linksys's website and click on partners then apply for
partnership you will be able to get access to the documents programs
you need
Hi,
I'm trying to have a SER machine send calls to an Asterisk server
working as an IVR. I was able to do this part just fine. Also, when
the caller makes certain options in the IVR, the call is then
transferred to an extension via SER. This part is also just fine.
However, I'm trying to
1 - 100 of 589 matches
Mail list logo
