Its not that the answers aren't out there, nobody bothers looking for them.
Being a relative new-b on Asterisk, I have to agree most of the
information is available on several Internet pages. However,
information is scattered around on many pages, and for someone who
isn't familiar with stuff
Hi all,
are you able to see incoming calls at the isdnlog? I have guessed I have a
problem
with the capi/isdn/card itsself and not really with asterisk.
Felix
Thanks I will give that a try.
Looks like this may need a bug report? We are all getting the
same errors.
Outgoing is fine
Welcome to the Asterisk users community!
Asterisk.org is a fast moving project. New code is added every day.
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Our community is also growing
As I said, I'm not an expert, so I would strongly recommend against
committing this as-is... someone please interpret why this works and
fix the root problem (or help me understand why this works so I can fix
the root problem).
I'd suggest that you open a bug with you problem and your patch
I was wondering if anyone knew of a way to create a busy-redial feature in
the * dialplan? For example, you try to call 12125551212 but the number is
busy, so you hang up and dial *XX12125551212 and hangup again, then * would
continue to retry calling the number until either it rings or a timeout
try to vi the app_prepaid.c and there is a line #include
postgresql/libpq.h and edit it and make #include libpq.h this problem
exist in modifyed app_prepaid only.
Best Regards
Hekuran Doli
Eureka... i must edit the Makefile hehe... yes correct, i must mantion
about the psql lib
-L
Jeremy McNamara wrote:
Michael Manousos wrote:
The performance of the oh323 channel driver is limited by OpenH323.
asterisk-oh323 uses the (more complete) RTP implementation offered by
the library, and not that of Asterisk. Of course there are pros
(adaptive jitter buffer, RTCP implementation)
Jim Rosenberg wrote:
--On Monday, June 28, 2004 7:21 PM +0200 Michael Sandee
[EMAIL PROTECTED] wrote:
Other than that... if these problems are not being published when
fixed... then other distro's do not have a chance to fix it... (think
about distro's that use stable code, but haven't updated
Florin Andrei wrote:
On Mon, 2004-06-28 at 07:45, Michael Manousos wrote:
Hello all,
Bugfix release 0.6.3 is now available. Basically, call indications
should work ok now. Also, the OH323 channel variables for incoming calls
are set properly (they can be used for special authentication purposes).
Hi,
-Original Message-
I was wondering if anyone knew of a way to create a
busy-redial feature in
the * dialplan? For example, you try to call 12125551212 but
the number is
busy, so you hang up and dial *XX12125551212 and hangup
again, then * would
continue to retry calling
Can anyone confirm if they have this channel bank running on Asterisk?
There is a post in the archive about having a few niggley problems but
no follow up.
I tried to email the guy but the mail address is bouncing.
I'm not having much luck finding any of the archive recommended channel
banks
Anyone had any experience here on how to config both ends, asterisk and
the sipura SPA1000
TIA
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Hi All
trying to compile asterisk under linux kernel 2.6.6.
Currently under zaptel get the following error
make linux26
Link /usr/src/linux-2.6 to your kernel sources first!
make: *** [linux26] Error 1
as going from the readme.
is 2.6 not compatiable with asterisk and should I go back to 2.4.26.
Jean-Yves Avenard [EMAIL PROTECTED] wrote:
I've only been watching this list for the past 2 days.
And it seems to be an one way street:
-Tell about your problems and what you would like to do.
Usually no answer.
I have to admit I'm rather disappointed with Asterisk, information is
On Tue, 2004-06-29 at 17:43 +1000, yaboo wrote:
Hi All
trying to compile asterisk under linux kernel 2.6.6.
Currently under zaptel get the following error
make linux26
Link /usr/src/linux-2.6 to your kernel sources first!
make: *** [linux26] Error 1
as going from the readme.
So ln
yaboo [EMAIL PROTECTED] wrote:
trying to compile asterisk under linux kernel 2.6.6.
Currently under zaptel get the following error
make linux26
Link /usr/src/linux-2.6 to your kernel sources first!
make: *** [linux26] Error 1
Type this:
# cd /usr/src
# ln -s linux linux-2.6
Thanks for the responses.
I have tried it with aggressive cancellation both on and off. I think that
on helps a tiny bit. I'm glad that Mike Benoit as seen something similar,
but of course sorry that he is suffering like me!
It is worse when I have a
Hi Tomaz,
make sure you disable the G723.1 codec in your SIP device, asterisk
does not support G723.1. Use G711 (alaw, ulaw)!
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
I find that if I drop the RX gain too much I start to lose DTMF decoding.
The Asterisk calls lose at least 3-6db end-to-end compared with a normal
call. If I bring the gain up, the symptoms sound exactly like yours.
The gain I am using is more like Rx=-2, Tx=0 but this is still quite quiet.
I
Ok I'm no cvs expert is there a cvs command to get a date sequential cvs
log archive for cvs-head or a URL for it. With so many daily changes its
hard to keep track of what the changes are.
Thanks
Chris
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[EMAIL
Hiya,
Looks like this may need a bug report? We are all getting the same
errors.
Sure, but i guess a bug to bugs.digium.com will be rejected, chan_capi
is not in CVS. Maybe [EMAIL PROTECTED] could do a fix, :-D PLEEASE :-D?
BTW: kapejod, any chances to disclaim chan_capi to digium? It would
Hi !
I need a solution to park incoming calls
to an extension of my choice where a special
announcement is played, park subsequent calls
to specific pools so that they listen to announcements
of my choice.
any ideas ?
Shah.
___
Asterisk-Users
Hi,
Would be interestd in anyones ideas for this problem..
We are starting a new division to our company, the people in this new
division will be the same people who are on the old division..
Calls for each division come in on seperate numbers and go through
seperate menus but ring to common
Jorge Mendoza wrote:
Hi,
I'm testing a Polycom IP600.
With firmware version 1.1 the phone reboots at any time.
With firmware version 1.2, the first reboot was an endless reboot. Then
I moved the phone to another lan port, then it worked fine. Then I
installed again in the initial lan port and
Can anyone think of any easier ways?
How about if you put second division on different server, and then share
VM storage on the network between two asterisk boxes?
SJ
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Ok I'm no cvs expert is there a cvs command to get a date sequential
cvs log archive for cvs-head or a URL for it. With so many daily
changes its hard to keep track of what the changes are.
You can use GUI tools like cervisia. Or use cvs2log script or so. Just
do your usual google searching
Senad Jordanovic wrote:
Can anyone think of any easier ways?
How about if you put second division on different server, and then share
VM storage on the network between two asterisk boxes?
SJ
The single server works fine for the two divisions making and recieving
calls..
Its that each
Hi all,
I want to use my * box to control entry to a building. I was wondering who else has
done this and what phones they might recommend.
The phone itself needs to be externally mounted so will have to be durable.
Functionally I would like it to just dial and extension when picked up.
Any
Ok I'm no cvs expert is there a cvs command to get a date sequential
cvs log archive for cvs-head or a URL for it. With so many daily
changes its hard to keep track of what the changes are.
You can use GUI tools like cervisia. Or use cvs2log script or so. Just
do your usual google
At 11:40 29/06/2004 +0100, you wrote:
Senad Jordanovic wrote:
Can anyone think of any easier ways?
How about if you put second division on different server, and then share
VM storage on the network between two asterisk boxes?
SJ
The single server works fine for the two divisions making and
I have a VOIP account including a telephone number in another
country.
The connection uses the H323 connection to connect to the
remote PBX.
Can I learn Asterisk to use that connection for outgoing calls
and also that he can handle my incoming calls?
Johannes
Jason Williams wrote:
At 11:40 29/06/2004 +0100, you wrote:
Senad Jordanovic wrote:
Can anyone think of any easier ways?
How about if you put second division on different server, and then
share
VM storage on the network between two asterisk boxes?
SJ
The single server works fine for the two
Hi! I'm trying to compile libiax2 on windows using
msvc6.
In the libiax2\src\iax.c file, line 670, I'm getting
a:
error C2229: struct __unnamed has an illegal
zero-sized array
It seems to complain due to the last member of
iax_frame. Does anybody knows what should I do to make
it compile?
Matt,
After much searching, I could not find any Ruggedised IP Phone's out of
the box...
I am now looking at using a Stunning Art Deco (:-) Dorfone (GBP300) from
www.spicecommunications.co.uk/shop/Door_Entry_Systems.htm
This range of speaker entry-phones, sit between your BT line and handsets;
I am now looking at using a Stunning Art Deco (:-) Dorfone (GBP300) from
www.spicecommunications.co.uk/shop/Door_Entry_Systems.htm
This range of speaker entry-phones, sit between your BT line and handsets;
when the buzzer is pushed the handsets ring.
I am guessing that this can be connected to a
Hi,
We've been working a lot with Asterisk in SIP for over 6 months but I've finally
succumb to the pressure of H.323. I need to find a way to do what we do with SIP but
with H.323. That is to have calls from H.323 peers placed into their own unique
context (unique to the endpoint placing the
Hello everybody,
we updated yesterday the full cvs version which include the h323
modification NoFastStart = TRUE in ast_h323.cpp So call to our GK EP are
again working. But we also connect to a gw which need FastStart. So
there, calls are still without audio.
Thanks for any hint
--
Daniel
On Tuesday 29 June 2004 02:42, Ralf Van Dooren wrote:
But instead of complaining about the lack of -findable- documentation,
one can try to enhance existing documentation. That's the power of
Open Source. As I am not a coder, I'll be trying to help the
community by making the documentation
tucker scribbled on Tuesday, June 29, 2004 2:37 AM:
Anyone had any experience here on how to config both ends, asterisk
and the sipura SPA1000
There is accurate information out there if you do a google search. The
first hit or two will have what you need. I don't remember if the
example I
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello
I'm the one who posted a message about the fact that nobody answer
anymore to questions asked.
I posted two days ago a problem I was facing that calls made over the
Internet to my Asterisk gateway would hang-up after just 5s (no NAT
were
Hi Tony-
Two E1's running on a TE405P should be a better choice, as the TE405 is
capable of bus mastering which gives better performance. Besides, you
only use one PCI slot in this configuration. However, of course you lose
board redundancy.
You can run 1 E1 port to another by using a
On Tue, 2004-06-29 at 04:43, Chris Stenton wrote:
Ok I'm no cvs expert is there a cvs command to get a date sequential cvs
log archive for cvs-head or a URL for it. With so many daily changes its
hard to keep track of what the changes are.
Go to http://lists.digium.com/ pick the link that
tucker scribbled on Sunday, June 27, 2004 8:56 AM:
This is what I want to do, however I am not sure how to achieve it,
can you help?
Asterisk is running with X100P card to local PSTN
Allow incoming calls over the internet to Asterisk
Allow internet calls to dial out (restricted) using
On Tue, 2004-06-29 at 05:42, Matt wrote:
I want to use my * box to control entry to a building. I was wondering who else has
done this and what phones they might recommend.
The phone itself needs to be externally mounted so will have to be durable.
Functionally I would like it to just dial
On Tue, 2004-06-29 at 08:36, Jean-Yves Avenard wrote:
I posted two days ago a problem I was facing that calls made over the
Internet to my Asterisk gateway would hang-up after just 5s (no NAT
were involved)
Answer I got was: it's your config
[snip]
So now which PCI slot your card is in
For outgoing calls made on our PRI circuit we are setting the Caller ID
using the format
Exten = _9XXX,1,SetCallerID(1601XXX)
The monitor shows that the CallerID is being set to the specified
number, but yet when the call is received on the user end the ID is
always the base number of
Jean-Yves Avenard wrote:
I posted two days ago a problem I was facing that calls made over the
Internet to my Asterisk gateway would hang-up after just 5s (no NAT were
involved)
Answer I got was: it's your config
Well, it wasn't (as I was expecting).
I compiled Asterisk under a Linux RedHat 9
Hi Folks,
I have the following situation:
I received an inbound call in my extension A and transferred it to the
extension B. But B was busy and I want to capture the call back to my
extension. How should I proceed?
Thanks,
Isamar
___
On Tuesday 29 June 2004 10:13, McInnis, JP wrote:
The monitor shows that the CallerID is being set to the specified
number, but yet when the call is received on the user end the ID is
always the base number of our DID. For example we have 8600-8650 as
DID's but the callerid is always 8600
Regardless of what you send in callerid, your PRI has a phone number
associated with it that you don't see, but is used for billing. This is
so you cannot spoof the LD company into thinking the call came from
somewhere other than from you. I believe the PRI provider can provision
the PRI to use
Drop the leading 1
On Tue, 2004-06-29 at 09:13, McInnis, JP wrote:
For outgoing calls made on our PRI circuit we are setting the Caller ID
using the format
Exten = _9XXX,1,SetCallerID(1601XXX)
The monitor shows that the CallerID is being set to the specified
number, but yet when
Hi,
On Mon, Jun 28, 2004 at 11:00:43PM +0200, Arve Rasmussen wrote:
What is the best SIP softphone to use with Asterisk?
Really don't know what is the best SIP softphone but I am
using linphone with alaw codec and dtmfmode rfc2833. Did not try
low bandwidth codecs until now.
we point people to the wiki
problem is that wikiware search sucks caterpillar snot, and
this particular wiki is a bit light on content and heavy on
links. one can spend massive time following links seemingly
relevant to a subject and never get to actual content about
it. often google yields
Some telcos require you only to send a certain number of digits. Try
sending fewer and fewer digits and see if it starts working. E.g.
instead of sending 1601abcdefg try sending 601abcdefg or abcdefg or defg
or even fg
If this doesn't work, from the CLI type pri debug span x and see what
you
I'm trying to do the following:
exten = i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
Any suggestions?
Thanks,
Isamar
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Hello,
When I enable SIP debugging I receive:
Peer RTP is at port 10.10.60.16:0
Shouldn't the RTP port be a number between 1 - 2?
- Brent
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[EMAIL PROTECTED]
DO NOT USE GROUP REPLY!!!
DO NOT USE GROUP REPLY!!!
DO NOT USE GROUP REPLY!!!
DO NOT USE GROUP REPLY!!!
DO NOT USE GROUP REPLY!!!
DO NOT USE GROUP REPLY!!!
DO NOT USE GROUP REPLY!!!
Got that...
DO NOT USE GROUP REPLY!!!
I will get a copy from the list server just
Hi Andrew,
I sympathise with your opinion. However if someone was
to analyse the messaged in the list they would find
that the most basic of questions get most replies. I
mean those questions that would take a few minutes to
answer searching through the wiki or google.
Where questions that are
Then it sounds like the Wiki is being mis/under-used. The great thing
about a Wiki is that if someone has an authoritative answer/solution to
a problem, they can just post it up.
Personally, I have found the Wiki to have plenty of relevant content
within its bounds; more esoteric topircs or
Hi,
Looks like your message got lost in the thread.
On 29/06/2004, at 8:47 AM, John Vogel wrote:
1. Use 4 Sipuras (approx. $400). Only problem is, I can't get this to
work! Sipura says use the G711 codec but it's not working for me.
Anybody have this working?
I haven't got any Sipura's to test,
DO NOT USE GROUP REPLY!!!
fat effing chance. fix your mail system. see a very large number
of threads. but as you seem unable to look up archives:-), try
this in your .procmailrc
# prevent dupes
#
:0 Wh: msgid.lock
| formail -D 65536 msgid.cache
Booo whh, use better
Patrick J. Conroy wrote:
Hello All,
I have finally pulled CVS HEAD and built it with app_rxfax and app_txfax to
try to solve the problem that I was having with blank faxes. Fortunately, I
am finally getting logs from rxfax. Unfortunately, I am still not receiving
faxes correctly. Here is the
Hi,
I'm playing around with Asterisk and DPH-100M (Dlink mgcp phone) on my
debian box. I've got stable version of Asterisk (packaged for debian)
working with dlink phone and 7910 from cisco (minimalistic
extensions.conf and chan_skinny for 7910) Everything works fine.
Now I'm trying to get
Title: Asterisk and dial-up modems
Anybody connecting to on-premise modems by dialing in to Asterisk and using an extension to reach the modem? How? Sipuras, FXO cards, other?
On Tue, 2004-06-29 at 09:53, Isamar Maia wrote:
I'm trying to do the following:
exten = i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
At that pint ${EXTEN} is i. Try using ${INVALID_EXTEN}
--
Eric Wieling * BTEL
Stephen R. Besch wrote:
snip
This is the install package for the program. Running setup will
install the program into Program Files\SachsLab\GSConfigure and put a
shortcut in the start menu under Phones. The sources are installed
to the application directory in a folder named Source. If you
On Tue, 2004-06-29 at 10:11, Randy Bush wrote:
DO NOT USE GROUP REPLY!!!
fat effing chance. fix your mail system. see a very large number
of threads. but as you seem unable to look up archives:-), try
this in your .procmailrc
# prevent dupes
#
:0 Wh: msgid.lock
I'm using version 1.9.1 build 3908
- next problem is that the text messages won't reach by another firefly
client
- Original Message -
From: Steven Critchfield [EMAIL PROTECTED]
To: Randy Bush [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Tuesday, June 29, 2004 7:31 PM
Subject:
From: Steven Critchfield [EMAIL PROTECTED]
To: Randy Bush [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
ROFL!
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On Tue, 2004-06-29 at 09:53, Isamar Maia wrote:
I'm trying to do the following:
exten = i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
Second time in 2 days, What extention are you at exten = i?
If you want to readback and invalid
I do. I decided not to bother with Vonage's sub-par and unmotivated
customer service(*) and plugged my ATA186 into an FXO port.
(*) Examples: Had three lines on two ATAs. Asked if I can moved one of
the lines off to a third (new) ATA -- they couldn't do it. Asked if I
can move an existing
never mind, new server + upgrades on the phones
sofware+ latest asterisk cvs = :)
- Original Message -
From:
Robb
Meredeth
To: [EMAIL PROTECTED]
Sent: Friday, June 25, 2004 6:59 PM
Subject: Re: [Asterisk-Users] chan_sip.c
max number of retries
One thing I
On 01:04 AM 6/29/2004, Florian Overkamp wrote:
Hi,
snip
busy, so you hang up and dial *XX12125551212 and hangup again, then * would
continue to retry calling the number until either it rings or a timeout is
reached, if it rings * then calls back the exten that made the *XX call and
bridges the
Maybe it is trying to say i as a digit?
You could have an [invalid] context with
[invalid]
exten = _.,1,Saydigits(${EXTEN})
and then include it at the very end of the [default] context (or wherever
you want to use it). That would then pick up anything that drops through. If
you do it any other
Hey guys, can you shead some light on
this?
I will copy my mgcp.conf and post below, but here
is the problem.
I can't get call waiting to work with my MGCP
device. I already have one call going, and I can hear the second call come
in, I flash over to it, but all I get is a dial tone, *
Title: Asterisk and dial-up modems
Look
at the ZapRAS 'show application ZapRAS' this only work w/a PRI.
TL
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of John
VogelSent: Tuesday, June 29, 2004 11:24 AMTo:
[EMAIL PROTECTED]Subject:
Jay Milk wrote:
I do. I decided not to bother with Vonage's sub-par and unmotivated
customer service(*) and plugged my ATA186 into an FXO port.
I never worked with vonage, is there tech support that bad?
--
Regards,
Steve Kalcevich,
Hello all-
This is probably easy, but I am trying to figure out why I cannot use app
record. In fact, the problem seems to extend to when using a sip soft phone
also. I always get this error, and haven't been able to find much about it.
I have a number coming in from nufone. I have tryied
Vonage does not allow any other device other than their own to be hooked up
to their system, period. There are whole bunch of service providers who
allow you to hook-up your own device. So why split hairs, use someone else
other than Vonage. Their is nothing extraordinary about Vonage, except they
hello,
i have trouble with nat + sip outgoing call.when make an outgoing call to a
sip gateway, i have no sound.
i have 2 sip gateway, one is asterisk.
asterisk is on public ip and private ip
other sip gateway is on public ip
phone are cisco and grandstream on private ip on the same subnet as
Hello
I have succesfully installed app_prepaidCID and populated the database. I
can send calls, but the only problem is that after call finish it does not
update the billing ballance on card. can any one help me about this?
Best Regards
Hekuran Doli
I wasn't able to get debugging information the first time around either.
After pulling the latest asterisk from CVS, I was able to build and see
debugging information when I started asterisk to test using
asterisk -vvgc. But I noticed today that I do not get the same
debugging information
Hi Everyone,
I'm one of those newbie users that really doesn't know what is going
on. And in stark contrast to the posts you may read, I do actually
think that when you post on this list, you get a reply. In fact, I know
you do. I have had a quite nice run setting up asterisk, when I've had
Eric Wieling wrote:
On Tue, 2004-06-29 at 09:53, Isamar Maia wrote:
I'm trying to do the following:
exten = i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
At that pint ${EXTEN} is i. Try using ${INVALID_EXTEN}
Eric,
Thank you, I've added that
You replied to a message with the subject of:
Re: Do people actually answer questions here?
And then changed the subject and started typing. This has wreaked havoc on
everybody's threaded readers, and made your question impossible to reply to.
You need to start a new message in your mail app and
Leif Madsen wrote:
On Tue, 29 Jun 2004 11:08:43 +1000, Jean-Yves Avenard
[EMAIL PROTECTED] wrote:
I have to admit I'm rather disappointed with Asterisk, information is
probably available but very hard to find ; it seems to be limited to a
few privileged people for whom their job is setting up VoIP
On Sun, 27 Jun 2004, Max Lock wrote:
Hi Folks,
Since there isn't a grandstream forum AFAIK I guess someone here may be able to
shed some light on this. Apologies if this is viewed as offtopic..
I think I may have killed the firmware on my Grandstream Budgetone 101. I found a
On 29 Jun 2004, Adnan Shah wrote:
Hi !
I need a solution to park incoming calls
to an extension of my choice where a special
announcement is played, park subsequent calls
to specific pools so that they listen to announcements
of my choice.
any ideas ?
Put up a bounty on it of a
Hi-
Perhaps someone with an E100P in hand can answer this:
I just received an E100P from Digium (I normally buy quad boards)
I noticed that the circuit board says T100P on it, and I assume that the
T100P and E100P both use the same circuit board.
Can someone please confirm that their E100P
hi,
is there a ldap-lookup for asterisk ??
i am seraching for asterisk app which i can give a name and a phone-type (sip,
iax, cellphone, work, home.) and get phonenumber back. And where i can
give a number and get the name.
Is there anything like that?
bye
Hi,
I'm trying to get asterisk to auto-dail out. I created a *.call file
with the the top of it being Channel: Zap/1/2609944, which should have
connected to Zap channel 1 and dial out to 2609944, but It did not do
so, asterisk would say a call was completed to Zap/1/2609944 but I never
heard
On Tue, 2004-06-29 at 05:10, Kevin Walsh wrote:
I'm using 2.6.7-gentoo-r6 and it works very well.
I assume you're using the latest Zaptel and Asterisk from CVS.
Do you have CVS ebuilds? That would make it a lot easier for us Gentoo
folks.
Thanks!
--
Robert Withrow, [EMAIL PROTECTED], +1 978
- Original Message -
From: Matt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 29, 2004 12:42 AM
Subject: [Asterisk-Users] Ruggedised IP Phone
Hi all,
I want to use my * box to control entry to a building. I was wondering who else has
done this and
what phones they
Like I said, they just seem to be lazy and/or badly organized. If they
can do LNP, why can't they change a hardline into a softphone, break
one number out onto a different ATA, etc? I basically laid it out for
them, saying If you can't move my 2nd line from this ATA to a new ATA,
then I'll need
Jorge,
That sounds strange to me. I have 12 IP 600s running without any of the
problems that you are having. My first guess would be that they are
configured wrong. Is the phone registering with asterisk? Is the phone
dowloading it's config files from the FTP server? If you want to post
your
[answeringsvc]
exten = 0,1,Wait,1
exten = 0,2,Dial(ZAP/g1/7135551212,${RING_TIMEOUT},r)
exten = 0,3,Dial(ZAP/g1/7135551212,${RING_TIMEOUT},mr)
exten = 0,103,Goto(0,3)
exten = 0,104,Goto(0,3)
This should call 713-555-1212. If there are no ZAP lines available it
should kick back around and play
On Tue, 2004-06-29 at 11:06, Doug Harris wrote:
Vonage does not allow any other device other than their own to be hooked up
to their system, period. There are whole bunch of service providers who
allow you to hook-up your own device. So why split hairs, use someone else
other than Vonage.
On Tue, 29 Jun 2004, Andrew Elchuk wrote:
I'm trying to get asterisk to auto-dail out. I created a *.call file
did you create the file in /var/spool/asterisk/outgoing/, or did you
create it elsewhere and then move it to that directory? The docs mention
that if the file is created in the
This sounds like it should be relatively simple to do in theory.
Couldn't you just create specific extensions that set the MusicOnHold
context (to play your different announcements) then transfers the call
to the parkext in parking.conf?
However, what do you want to happen when the
Webn1 a écrit :
hello,
i have trouble with nat + sip outgoing call.when make an outgoing call to a
sip gateway, i have no sound.
i have 2 sip gateway, one is asterisk.
asterisk is on public ip and private ip
other sip gateway is on public ip
phone are cisco and grandstream on private ip on the
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