On Sat, 2004-09-11 at 14:25, Kevin P. Fleming wrote:
Adam Goryachev wrote:
PS, in case you are wondering, I (and my supplier) have spent hours
looking at different motherboard specs, and so far haven't been able to
find anything suitable (except a dual opteron motherboard and just using
Chad Brown wrote:
How do you configure an outbound proxy for Asterisk? If the sip call is
not local I want everything to go to a designated sip proxy.
In the standard chan_sip, there's no support for outbound proxy.
In my chan_sip2 test channel, I have that support. Please test!
If I get enough
Does this mean i will need a vocoder card to make this work?
The vocoder card has the ability to do both alaw and ulaw 64 codecs ala
asterisk
I am attempting to connect an asterisk system into an existing Network
Alchemy branch. This system supports h.323 and has an optional vocoder card
I fixed the problem now. There was something wrong with our provider.
On Fri, 10 Sep 2004, Sean Lowry wrote:
What alarm is it. Is it red or is it yellow.
If it's red then it's the /etc/zaptel config
But if it's yellow then it's a problem with sync the channels
Which could be a master -
hi all,
I am running a small asterisk box at home and i love playing with
it ..
I have come across the option canreinvite=yes which will
connect the media path
directly . I have used this sucessfuly ..
What will happen if one of the phones hangs/crashes during the
call .. i mean, what
Larry,
I have this setup currently
Win - * - Linux - Internet
^
|
Win|
I am using iptables FW on linux just to browse
internet which has two NICS on it.
All the setup is internal network. May be it is
creating issue?
Every help will take me step ahead.
thanks
dev
--- Larry
On Tue, 14 Sep 2004 14:23:54 +1000, Phil Stevens
[EMAIL PROTECTED] wrote:
I don't quite understand what you need Austel compliance for if all you
want to do is link to a PBX. You don't need Austel approval for that,
only if you connect directly to Telstra.
Can anyone please confirm this?
Hi everyone!
I now have obtained a couple of SIP-accounts at a local VOIP-provider.
How do I specify that ALL outgoing calls to _NXXX go out via one of
these accounts?
Regards,
Evert
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To check to see if it's registered and to setup the
correct busy and unavailable messages I have a macro which checks this for
me. This works for Zap and sip.
[macro-stdexten]exten =
s,1,Dial(${ARG1},20)exten = s,2,Voicemail(u${MACRO_EXTEN})exten
= s,3,Goto(incoming,s,1)exten =
You can try this:
In your sip.conf add the following entry
[yourProvider]
type = peer
secret = yourPassword
username = yourUsername
host = yourProvider
fromuser = yourUsername ; some prviders need this parameter
fromdomain = yourProvider ; some prviders need this parameter
In your
Chad Brown a crit :
It looks like chan_sip2 may solve my problem with outboundproxy
support. However, I am having problems getting the solution installed.
From what I understand these are the tasks
Add chan_sip2 to the channels/Makefile
* Rename the file downloaded to chan_sip2.c
* make / make
administrator tootai a crit :
[...]
No. You have to add also a
chan_sip.so: chan_sip.o
ifeq ...
Ooups, my pardon, should be:
chan_sip2.so: chan_sip2.o
ifeq ...
You can copy/paste this from chan_sip and adapt it. Please take a look
on chan_sip2 page, there is a makefile sample (or diff, don't
Hi all!
I'm trying to have asterisk route all outgoing calls out via my VOIP
provider.
exten = _NXXX,1,Dial,SIP/[EMAIL PROTECTED] seems to have them to in
the direct direction. However, debug shows that my asterisk doesn't
identify itself correctly:
Sip read:
SIP/2.0 100 Trying
From:
I have tested alaw and it works fine with no vocoder! (argent branch -
asterisk)
problem solved.
Stuart.
The vocoder card has the ability to do both alaw and ulaw 64 codecs ala
asterisk
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Hello List!
I finally got asterisk with capi working, and its already answering my
call as well! :)
Now i would like to call a number from my shoft phone (kphone).
This is my extentions.conf:
---
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ;
From a 'sip debug':
Sip read:
SIP/2.0 100 Trying
From: Evertsip:[EMAIL PROTECTED] ext. IP];tag=as6e18534e
To: sip:[dialled [EMAIL PROTECTED] server of VoIP provider]
Call-ID: [EMAIL PROTECTED] ext. IP]
CSeq: 102 INVITE
Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b
Content-Length:0
7
I would have said you need something like this in extensions.conf
Dial(CAPI/yourmsnnumbergoeshere:${EXTEN:1})
And nothing in modem.conf
And in capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
mode=immediate
controller=1
softdtmf=1
accountcode=
Tried that. Now I get:
Sip read:
SIP/2.0 403 Forbidden (From header is not a Trust host or gateway)
From: Evertsip:yourUsername@yourProvider;tag=as0687982f
To: sip:069101701@yourProvider;tag=87f2a0d5-13c4-4146e66c-1a8baa18-5e5
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Via:SIP/2.0/UDP
Hi,
I'm new to *. I just installed my X101P card with * from the source on
Mandrake 10.0 and I test it. Everything seems to work fine. When I call
at my home office all the demo ivr seem to work. But I have one question
regarding * using /dev/dsp0. I only have one sound card on my system and
it
On 14 Sep 2004 at 12:04, David Davies wrote:
I don't have a dev box to hand that I can test with but
If you change it to something like
ignorepat = 9
exten = _9.,1,ResponseTimeout(100)
exten = _9.,2,Dial(CAPI/35:${EXTEN:})
You should be able to dial 937 to connect to 37 ?
I think that
on the asterisk site, it was stated while ago, how to download stable
version. like
cvs checkout -r v1-0_stable asterisk-addons zaptel libpri
but now it's not their. is stable-version removed from the CVS ?
or is their some different procedure ?
thank you
--
Atif
I think you should install the openssl and openssl-devel packagesDinesh [EMAIL PROTECTED] wrote:
cd ../asterisk# make clean; make installHello when I do a make clean and make install, I get this error message onmy asterisk box.bdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv
I've successfully got inbound/outbound calling working with our Asterisk
using the Asterisk-OH323 channel driver. We are using a parent gatekeeper
and the NuFone H323 channel driver would not work with the parent
gatekeeper...
I'm trying to determine a way to ensure that the line used for outbound
I need a cheap platform for installing a tdm400. Could someone tell me if the cheap cpubuilders computer at sams $179 (cbs110l) is pci 2.2 compliance? I ve got a compaq deskpro en 700 that does not seems to be compliant and I need to change it to start developing. Thanks for the help. Computer
Have you tried to register directely to you VoIP provider using a soft
phone (such as
X-Lite) to check that your account (user/passwd) is activated ?
You can also try different values for the fromdomain parameter ...
On Tue, 14 Sep 2004 12:45:07 +0200, Evert Meulie [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
From what I can tell when I place an outbound call from
Asterisk it always tries to use the first registered H323
alias...
My dial plan in extensions just says Dial(OH323.)
Unlees the gatekeeper rejects multiple calls from Asterisk,
there's no need for multiple
OH323 registers itself as a Gateway, and the H323 channel as a terminal.
Afaik there is no easy way to change it.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas
Sikkema
Sent: Tuesday, September 14, 2004 9:30 AM
To: Asterisk Users Mailing List -
I had the same problem under Red Hat Fedora Core 2.
Try this:
uname -r(will get you the 2.6.5-1.358, in my case)
And then:
ln -sf /lib/modules/2.6.5-1.358/build/ /usr/src/linux-2.6
ln -sf /lib/modules/2.6.5-1.358/build/ /usr/src/linux
Go to the /usr/src/zaptel directory as root and
The only disadvantage we found to using the OH323 channel driver is that we
cannot now register netmeeting or other h323 directly to the * With the
nufone h323 channel driver we could register netmeeting and other h323
devices directly to the *...
So if we wanted to run internal h323
On Tue, 14 Sep 2004, Evert Meulie wrote:
Hi all!
I'm trying to have asterisk route all outgoing calls out via my VOIP
provider.
exten = _NXXX,1,Dial,SIP/[EMAIL PROTECTED] seems to have them to in
the direct direction. However, debug shows that my asterisk doesn't
identify itself
Hi,
I'm thinking of introducing Asterisk on Linux for IP PBX.
Now I'm using ISP that has VoIP service and I have VoIP terminal box for
that ISP and a SIP account for SIP server of the ISP.
Now, what I would like to do is the following.
A. Setup IP PBX on Linux by
On Tue, 14 Sep 2004, Kuniyoshi Murata wrote:
I'm thinking of introducing Asterisk on Linux for IP PBX.
Now I'm using ISP that has VoIP service and I have VoIP terminal box for
that ISP and a SIP account for SIP server of the ISP.
Now, what I would like to do is the following.
A. Setup IP
Felix,
You might try going to Sam's (or their website) and looking
for the motherboard manufacturer on their marketing materials. Then you
can get the specs for the motherboard from the mobo maker. I cannot
imagine anyone here will know if the PC you are reference is compliant of the
tops
On Tue, 14 Sep 2004 21:48:52 +0900, Kuniyoshi Murata
[EMAIL PROTECTED] wrote:
A. Setup IP PBX on Linux by using Asterisk.
B. For IP-PSTN gateway, configure Asterisk to use my ISP's SIP account and
connect to my ISP's IP telephony service.
Is that possible?
Murata-san,
A is certainly
I have had many conflicting conversations with Mitel dealers, resellers and
executives, they don't seem to know what they want to do. At one point they
say they will be releasing a full line of easily configurable SIP phones,
then they take that back, then they say they will be backward compatible
On Tue, 14 Sep 2004 06:47:53 -0400 (EDT), Wayne Veilleux
[EMAIL PROTECTED] wrote:
Hi,
I'm new to *. I just installed my X101P card with * from the source on
Mandrake 10.0 and I test it. Everything seems to work fine. When I call
at my home office all the demo ivr seem to work. But I have one
Title: SIP call server- Too many hops
I have Asterisk server setup
Two SIP phones register successfully
When SUBSCRIBE or INVITE messages are sent to the SIP server, it returns Too many hops
Any ideas?
I am using the domain name on the clients to be the IP address of the SIP server
Hmmm
Thank you Eric.
Do you know where can I see all ${DIALSTATUS} available?
-Mensagem original-
De: Eric Wieling [mailto:[EMAIL PROTECTED]
Enviada em: segunda-feira, 13 de setembro de 2004 11:22
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Assunto:
Hey all,
I'm trying to get my Asterisk server up and running on
fwd.pulver.com just to get the hang of it until I get
my FXO card in a couple of days. It seems to connect
but that's about it. If I try to dial into it from
another fwd # it says user is not online.
In sip.conf I have the
On Tue, 2004-09-14 at 05:41, Dinesh wrote:
cd ../asterisk
# make clean; make install
Hello when I do a make clean and make install, I get this error message on
my asterisk box.
bdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv -lssl
/usr/bin/ld: cannot find -lssl
README.variables I think in the Asterisk docs directory.
On Tue, 2004-09-14 at 09:08, Jozeph Brasil wrote:
Hmmm
Thank you Eric.
Do you know where can I see all ${DIALSTATUS} available?
-Mensagem original-
De: Eric Wieling [mailto:[EMAIL PROTECTED]
Enviada em: segunda-feira,
All
Probably teaching you all to suck eggs but.
My provider is Kingston Comms ( UK ) and I have had a bit of a struggle to
get the * system setup on their cct's.
Just thought I would let you all know what I had to do.
Firstly order ISDN 110 they will try to provide ISDN 85 as standard.
Make
Hello everyone,
I am having big problems trying to detect dtmf
tones while a IVR prompt is playing on zap channels. Sometimes the detection
only starts 4-5 seconds into the prompts. Other times it works very well for the 1st few calls and then starts having problems.
And most times it also
Could you check to see if both your SIP clients are registering with
Asterisk? Issue the following cmd from the CLI:
*CLI sip show peers
If your clients are properly registered you should see a similar output.
Name/usernameHostDyn Nat ACL Mask Port
Status
Mario,
I solved this problem making teh following symlink :
ln -s /lib/modules/2.6.8-1.521/build/ /usr/src/linux-2.6
I hope this may help you.
Regards,
Juanjo
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Has anyone compared * to sipXpbx? From a cursory look, this open source
version of PingTel's PBX has many features that make it more suitable
as a replacement for a traditional PBX, including the ability for users
to tell if a phone/trunk is in use. What I am trying to figure out is
what I'd
Hello,
I've been using a Polycom IP 500 behind Asterisk for several months
now with fairly nice results. Just for grins today I decided to
register it directly against Broadvoice, and I noticed a particularly
fascinating phenomenon. When calling other Broadvoice users, upon
placing the call, my
Thanks Marconi, that solve my problem.
Bye.
Wayne
On Tue, 14 Sep 2004 06:47:53 -0400 (EDT), Wayne Veilleux
[EMAIL PROTECTED] wrote:
Hi,
I'm new to *. I just installed my X101P card with * from the source on
Mandrake 10.0 and I test it. Everything seems to work fine. When I call
at my
Hi,
Been using asterisk for a little while now, but were
starting to notice that a number of voicemails left have very broken sound. Were
using a Digium Quad E1 card, we had some problems with broken audio at first
that turned out to be a problem with interrupts, however those issues
I previously reported a problem with receiving inbound calls
with Galaxy Voice and the Current CVS Head version. I am currently running version 5/17/04 of the CVS HEAD and
my incoming calls work correctly with Galaxy Voice. If I upgrade to the Stable release or
the current CVS head I am no
On Tue, 14 Sep 2004 10:15:01 -0400 (EDT), Jon Miron [EMAIL PROTECTED] wrote:
Hey all,
I'm trying to get my Asterisk server up and running on
fwd.pulver.com just to get the hang of it until I get
my FXO card in a couple of days. It seems to connect
but that's about it. If I try to dial
Anyone have success with this option? Trying to use it in the following
scenario:
4 different managers call into the conference and all 4 hear music waiting
for the Exec to login
Exec calls into conference and all 5 people can hear eachother and talk to
eachother.
Exec has the ability to
On Sep 13, 2004, at 11:32 AM, Andrew wrote:
I'm a bit new to the terminology. Let me ask my question more simply,
even though I think you already answered that it should
work
I want to receive calls into the Asterisk PBX via a cheap POTS-PBX
method, such as a WinModem or other FXS endpoint on
Carmi Weinzweig wrote:
Has anyone compared * to sipXpbx? From a cursory look, this open
source version of PingTel's PBX has many features that make it more
suitable as a replacement for a traditional PBX, including the ability
for users to tell if a phone/trunk is in use. What I am trying to
Are you referring to GrandStream? If so, I sent them an e-mail asking about
the new 102D model and bellow is their response:
+ Dear Iassen,
+
+ The 102D will be available in November/2004. It will support multi-line,
+ conference call, headset jack, speed dial keys and other great features.
+
+
Hello list,
I have upgraded the queue_log analyzer I was talking about last month so
that some sample pages are available in English too, and this time I
tested them with Firefox too. :-)
This is mostly an incremental upgrade meant to show what the program is
currently doing. I expect to
I have 2 problems with openswitch12:
1)
I can not make work "ignorepat = 9" i do not get dialtone after the
number is dialed, the system ignore the number and i can go on dialing
the rest of the number but when i want to take the line teh
dialtone do not stay.
2)
when i tray to leave a
Apparently there are subtle diferences between meaning of MeetmeJoin
event depending on channel type.
Problem is: after originating a call from channel to MeetMe room i.e.:
[meetme]
exten = 1,1,Answer
exten = 1,2,Meetme(kolejka|dqM)
than:
Context: meetme
Exten: 1
Priority: 1
ActionID:
On Tue, 2004-09-14 at 11:30, Maciek Kaminski wrote:
Apparently there are subtle diferences between meaning of MeetmeJoin
event depending on channel type.
Problem is: after originating a call from channel to MeetMe room i.e.:
[meetme]
exten = 1,1,Answer
exten = 1,2,Meetme(kolejka|dqM)
For almost 6 months now I've upgraded Asterisk every couple of weeks or
so and I've never had this problem. When I'm at the asterisk console
(asterisk -r) it shows me live status. Who called who, what it's playing
and when, etc. It logs to the screen. When I type reload, it says added
so and
Is your asterisk started with no -vvv options ...
Julian
- Original Message -
From: Deon Rodden [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 14, 2004 5:58 PM
Subject: [Asterisk-Users] Asterisk not outputting real time display
For almost 6 months now I've upgraded
Hi,
But recently I upgraded and now when I do reload all I see is Sep 14
12:55:25 NOTICE[393230]: indications.c:397
ast_unregister_indication_country: Removed default indication
country 'us' and nothing more. What's worst, status is not
being updated, I can't see who calls who and if
When I do a 'asterisk -vc' I get following, but asterisk does NOT stay up:
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
I was just about to replace 2 Merlin Legend systems (one in my house
and one in my Parent's house) with * systems. I have 2 beefy linux
systems ready to go and have enough Cisco 7960 and 7970 phones to
replace all my MLX-10D, MLX-16D and MLX-20 phones. Then I discovered
that * seems to be
On Tue, 2004-09-14 at 11:32, Prof. Marcelo Kruk wrote:
I have 2 problems with openswitch12:
1)
I can not make work ignorepat = 9 i do not get dialtone after the
number is dialed, the system ignore the number and i can go on dialing
the rest of the number but when i want to take the line
Hello!
Am Dienstag, 14. September 2004 19:21 schrieb Evert Meulie:
Found new ID3 Header
Remove the ID3 Tag from your Music-On-Hold MP3s with:
http://fibiger.org/mp3tag.html
Andi
--
- Andreas Roedl- Senior IT Manager
- NATIVE INSTRUMENTS GmbH - [EMAIL PROTECTED]
- Schlesische
snip
so you have ACD on the AVAYA? are you thinking of front ending all calls
to the ACD with the IVR? I would think that you would set the IVR up as
an option within the ACD instead of front ending all calls to the ACD.
The flexibility of * is great for this and I have built several IVR's for
Hello everyone,
I am having big problems trying to detect dtmf
tones while a IVR prompt is playing on zap channels. Sometimes the detection
only starts 4-5 seconds into the prompts. Other times it works very well for the 1st few calls and then starts having problems.
And most times it also
With the SNOM 220 and its 20 additional LED/buttons... how can we light
a button when the extension is in use elsewhere?
Thanks,
Carmi Weinzweig wrote:
I was just about to replace 2 Merlin Legend systems (one in my house and
one in my Parent's house) with * systems. I have 2 beefy linux
Ok, ok, I know there
has been plenty of discussion on asterisk and fax - from this I
understand:
1)
First and foremost, use g.711 ulaw
2)
Packet loss, etc...makes faxing over the internet unreliable
My need is for a fax
to come in on a X100P and be forwarded to a fax machine on the local
On 2004.09.14 11:10 Marty Mastera wrote:
Ok, ok, I know there has been plenty of discussion on asterisk and fax
-
from this I understand:
1)First and foremost, use g.711 ulaw
Yes, the codec must be lossless.
2)Packet loss, etc...makes faxing over the internet unreliable
I'm not sold on
http://bugs.digium.com/bug_view_page.php?bug_id=0002379
Updated to try and fix the issue others were seeing. Added a check in front
of ast_frfree(f); Just to make sure we do the right thing(tm)
bkw
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The other thing you'll probably want is a insecure=very in your [fwd]
section of sip.conf - This caught me out the other day, but was documented
on the wiki.
Cheers
Paul
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I am recieving inbound calls to my asterisk box over IAX.
And getting spawn extensionexited non-zero errors.
Im not entirely sure what this means, and would appreciate any help in
fixing my problem.
FYI:
** is the inbound phone number
x.x.x.x is a remote asterisk box calling my own
Greetings,
I've just gotten asterisk working such that it will accept incoming calls
and play the demo recording (from make demo) during the install.
However, on the console of the machine asterisk is running on, I'm getting
sound recording overrun errors.
The only other indicator I'm getting
I usually use safe_asterisk or /etc/init.d/asterisk start the
defaults have always worked for me.
Nick Barnes wrote:
Hi,
But recently I upgraded and now when I do reload all I see is Sep 14
12:55:25 NOTICE[393230]: indications.c:397
ast_unregister_indication_country: Removed default
To answer my own question:
I was told on IRC that the IAXy doesn't renew it's lease.
So I'm going to make the lease permanent.
gat
Glenn A. Thompson wrote:
Hi,
I have an IAXy which *appears* not to renew it's DHCP lease.
The DHCP server is a Solaris box running the native Solaris DHCP server.
Is
Hello,
I upgraded to cvs-head over the weekend and now agents that are logged
in on zap channels have to acknowledge ACD calls by pressing #, even
though I have
ackcall=no
in agents.conf. This doesn't seem to be happening to agents on SIP
phones, and it this is the first time I have had an
i am in a macro. ${ARG2} is a phone number, which might be
seven, ten, or eleven digits. i wish to canonicalize it to
be a full 11 digit number. if this was a normal exten, i
would
exten = _1XX,1,GoTo(dial-gateway,${EXTEN},1)
exten = _XX,1,GoTo(dial-gateway,1${EXTEN},1)
On Tue, 2004-09-14 at 14:25, Randy Bush wrote:
i am in a macro. ${ARG2} is a phone number, which might be
seven, ten, or eleven digits. i wish to canonicalize it to
be a full 11 digit number. if this was a normal exten, i
would
exten = _1XX,1,GoTo(dial-gateway,${EXTEN},1)
I have been trying to locate the patch that is supposed to cure the
problem of hearing sound from the previous call when dialing through i4l
and an hfc card. Does anyone have it? It is mentioned briefly in this post:
http://lists.digium.com/pipermail/asterisk-users/2003-February/007530.html
I'd be grateful for some help on this. I've been following the various e-mails on the
UK CID issue, particularly the last posting in bug fix 9.
It seems that all I need now is to apply ast-UK-and-DTMF-pol-CID.diff.
I apologise in advance for what is probably a very simple question, but how do
A potential reason for the difference could be if Asterisk uses UDP (which I
think that I've read somewhere it does). TCP is a protocol that demands
that transmitted packets are numbered and that the receiver both request the
re-send of packets that appear to be missing and order any packets that
Hello List!
I have this simple problem:
fileserver:/usr/src# capiinit start
modprobe: Can't locate module kernelcapi
ERROR: cannot load module kernelcapi
Although, i think i hav everything build in, and loaded:
~# grep -i capi linux-2.4.27/.config
CONFIG_ISDN_CAPI=y
Guys:
I routinely run multiple phones over our satellite system
(I'm the VP of Network Services at SDN Global, a satellite bandwidth provider
located in Charlotte NC, US).
Just last week I went to West Palm Beach, FL USand
turned up a 10 phone emergency call center, complete with ACD
I noticed that the linux version of Cepstral is now available. however
its name is now swift, not theta.
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To UNSUBSCRIBE or update options visit:
You need to run:
patch -p1 ast-UK-and-DTMF-pol-CID.diff
You may need to change the -p1 to -p0 depending on the paths in the diff.
I don't think this patch applies cleanly with current CVS head - I know for
sure the 31-08 version does.
HTH
Ed
-Original Message-
From: [EMAIL PROTECTED]
Hi Matt,
There could not be a more timely response, It seems to have worked for me.
added chanspy.so to the make file.
thanks
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matt G
Sent: 14 September 2004 21:25
To: Asterisk Users Mailing List -
My take on this is that Mitel doesn't know internally how they are going to
play out interoperability since they have a vested interest in keeping users
locked in with MiNet. Of course, lock-in is anathema to the *whole point* of
a VoIP system, so this, to me, is a stupid business decision and the
Tim McKee wrote:
Guys:
I routinely run multiple phones over our satellite system (I'm the VP of
Network Services at SDN Global, a satellite bandwidth provider located in
Charlotte NC, US).
Just last week I went to West Palm Beach, FL US and turned up a 10 phone
emergency call center, complete
I need to implement a procedure for creating a 3-way call, similar to what
you get from the telephone company. You're in a call, you flash hook to get
the switch's attention, you dial the 3rd party, you flash again to create
the 3-way call.
In the asterisk world, the flash would be replaced with
On 14 Sep 2004 at 12:04, David Davies wrote:
I think that should be:
ignorepat = 9
exten = _9.,1,ResponseTimeout(100)
exten = _9.,2,Dial(CAPI/35:${EXTEN:1})
(notice the exten:1 - the 1 represents how many digits to strip.
Since you want to dial 937, you want to strip the first digit,
The voip wiki says that three way calling is implemented on the client side.
I searched google and found some previous threads on the Asterisk list that
may be helpful to you:
http://www.voip-info.org/wiki-Asterisk+PBX+functions
On Tue, 2004-09-14 at 17:04, Bill Hamlin wrote:
I need to implement a procedure for creating a 3-way call, similar to what
you get from the telephone company. You're in a call, you flash hook to get
the switch's attention, you dial the 3rd party, you flash again to create
the 3-way call.
That works exactly as expected on Zap interfaces. For VoIP devices it's
TOTALLY handled by the phone.
If you're extremely lucky :)
-Chris
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On 14 Sep 2004 at 18:20, Matt Williams wrote:
I am recieving inbound calls to my asterisk box over IAX.
And getting spawn extensionexited non-zero errors.
Im not entirely sure what this means, and would appreciate any help in
fixing my problem. FYI: ** is the inbound phone number
On 15 Sep 2004 at 1:52, San Singhania wrote:
Hello everyone,
I am having big problems trying to detect dtmf tones while a IVR
prompt is playing on zap channels. Sometimes the detection only starts
4-5 seconds into the prompts. Other times it works very well for the
1st few calls and then
On Tue, 2004-09-14 at 10:34, Phil Stevens wrote:
Hello,
I'm researching the possibility of using VOIP (SIP) with an existing
PABX system. Ideally, the setup would be to dial an outside line through
the PABX (that would actually link to the the VOIP gateway).
At this point I would prefer
On Tue, 2004-09-14 at 13:56, bagattin jerome wrote:
Hi,
Another question about read command:
Howto sup file option and keep maxdigits options ?
exten = 3,1,Read(ILE,1)
give me :
Unable to open 1 (format UNKN): No such file or
directory :-(
This one is easy, just do:
exten =
Hello,
I have an E1 connected to an * server, which takes incoming calls and
verifies the existance of the called number in our internal E164 tree.
Now there is a number that exists on one of the servers, but the phone
has registered itself, so the dial plan executes an hangup. This hangup
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