Andrei (MPI) wrote:
Richard Scobie wrote:
It is a simple one liner.
...
Index: wctdm.c
...
+ reset_spi(wc,card);
...
This is exact same patch that Digium support tried before sending me new
fxo modules. That wctdm.c patch did not help in my case.
Interesting, thanks Andrei. I have run
Hi Steven,
I wish, I already have 2 spare TE410p and 1 TE405p. But customer wants to use
Analogue and they already installed the lines.
Yesterday, I ready about the FXO modules being replaced by Digium, this relaxed
me a bit.
But as you said, I will have to worry about the ring and hangup
Hi timebandit,
I'm realy happy to hear that, and as a matter of fact, all my Asterisk hardware
is Intel server products, from chasis to MB. I know I can trust this hardware
and I have excellent support from the guys I buy from.
Thank you.
Message: 5
Date: Mon, 3 Jan 2005 15:48:47 -0500
From:
I have a couple of small machines (home PBX) and run a (tiny) USB APC UPS
(nice set of TLA's!)...and love it. Serial ports are so 90's! :- I am
running:
Debian Linux
X100P FXO card
...and apcupsd:
http://www2.apcupsd.com/
(on debian apt-get install apcupsd)
I like it because it can slave 1 (in
I did post this a couple of days ago, but no-one has replied yetso I
thought I'd make the subject line a bit more accurate, add a bit, and try
again!
I've started looking at Asterisk as a possibility for a small PBX here.
I'm thinking of an ISDN (BRI) card for connection to the telco, with
Hello,
Sorry if this it totally OT.
Has anyone used the Linux version of CodeWarrior to build and deploy an
asterisk distribution ?
From what I have gathered, you can build an install image of a Linux
system using PCS, along with
the components you choose.(I may be wrong though)
Any comments ?
Not yet solved !
Serge Rodrigues
Research Development
* e-Mail [EMAIL PROTECTED]
* Tel +32 (02) 649 80 89 PERIACTES
* Fax +32 (02) 648 27 22 Av. de l'hippodrome 147
* Webhttp://www.periactes.be B-1050 Brussels -
Hello List :-)
I'm sorry that this is a bit off-topic, but I don't know where to ask this
question.
Is there anyone who can tell if Asterisk will be present at CeBIT this year?
Kind regards
Thilo
--
Thilo Rößler
Linup Front
Pallaswiesenstrasse 203
64293 Darmstadt
Tel: 06151/9067-0
Fax:
On Mon, 2005-01-03 at 14:34 -0700, Harry McGregor wrote:
Try finding a VOIP Phone that does gigabit... Won't happen for a while.
Cable is cheap when you look at the cost of running the cable. If you
use two boxes, it will take virtually the same amount of time to run two
as it does to run
cmould wrote:
Hi:
Just saw your post while trying to solve a similar asterisk problem.
Did not see any responses. Was your problem solved and what was the
solution?
well, i'm working on that problem for not a long time, before i was
doing something else, so i had little time to discover some
On 02/01/2005 at 11:21 Michael Graves wrote:
Hi All,
I've read J.R. Richardson's paper Create an Embedded Asterisk Server
which outlines making a Debian server that boots from a compressed disc
image on a CF card. I'm really interested in this as I want my * server
to be more like an appliance
What happens on the phone is that I hear voiptalk.org's
greeting and after they hang up, I hear my own invalid
extensions message.
That's because your _X. extension has a 2nd line. Reduce it to
_X.,1,Playback(invalid) and you won't have that problem.
I'm not the original poster but
I really am at my wits end about this one. Some people report this
card and server working fine while others (like myself) can't get it
going no matter what. I have been told by the Digium distributor in
our country that this card simply not compatible with some
motherboards. Sounds very weak for
Hi,
we have a couple of Snom 190 and want to get Intercom working with *.
The Wiki says something about utilizing an old firmware plus a small
change in chan_sip.
What are the chances to get intercom working with a Snom 190 with a
current firmware? Anyone? Is Snom working on this?
--
Best
I'm just about to start implementing this project. I have a test
server working well with SIP phones and IAX for incoming and outgoing,
but when I golive will need 4 analogue lines coming in.
1. Anyone got this config working with a 4 port FXO digium card
2. Any tips/hints/traps
Thanks
John
On Tue, Jan 04, 2005 at 10:13:27PM +1100, Eric Bishop wrote:
I really am at my wits end about this one. Some people report this
card and server working fine while others (like myself) can't get it
going no matter what. I have been told by the Digium distributor in
our country that this card
I created a tarball for redhat 7.3 with kernel 2.4.28, asterisk
v1.0.3-CVS-01.04.05 and alsa-1.0.6a... see the README / download it here:
ftp://ftp.linuxsys.com/pub/packages/rh73/asterisk/
--
Andrew McRory - President/CTO
Linux Systems Engineers, Inc. - http://www.linuxsys.com
On Mon, 3 Jan 2005, Race Vanderdecken wrote:
For the life of me I will never understand why people believe that each
cubical/desk needs it own 4-pair cable, CAT5 cable connect back to the
server.
One word: flexibility.
Actually, normally you would run 2-4 cables / desk at once. The cost is
I really am at my wits end about this one. Some people report this
card and server working fine while others (like myself) can't get it
going no matter what. I have been told by the Digium distributor in
our country that this card simply not compatible with some
motherboards. Sounds very
Hi
I want to know the best way to connect Asterisk to a Siemens HiPath HG1500
PBX. Until now I came out with 3 solutions:
1-Asterisk being a H.323 client of the Siemens PBX (I believe it needs
Siemens licences and Digium hardware)
2-Asterisk connecting to the PSTN phones with Voice Modems (good
I gave up on an HP 1U server (hmm, I think it was a DL 140)
It seemed like the E100 I tried to put in didn't play nice with the
server monitoring ASIC on the MB. I tried several distros and
kernels, and even installed HP's driver set for one (Suse I think).
Same card and config worked fine in a
Hi,
On Tuesday 04 January 2005 12:18, Peer Oliver Schmidt wrote:
we have a couple of Snom 190 and want to get Intercom working with *.
The Wiki says something about utilizing an old firmware plus a small
change in chan_sip.
What are the chances to get intercom working with a Snom 190 with a
hi
how can I configure the sample time period (10ms,20ms etc) for codecs?
roy
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Nils Ohlmeier wrote:
What are the chances to get intercom working with a Snom 190 with a
current firmware? Anyone? Is Snom working on this?
yes, Snom is working on this.
This is very good to hear. Do you have any time frame (2 weeks / 2 month
/ 6 month / 2 years)? Information for this is fairly
It seems the current issue is that the Snom phones in current firmware don't
want to accept intercom=true on the SIP URI. When passed this is the result
and Asterisk retries and retries before exceeding a maximum # of retries to
get the phone to connect.
SIP/2.0 401 Unauthorized
Via:
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
phone number on my phone. In the log is the following - which displayed
'601' on my phone. The caller was +886288097680 - am I getting the wrong
ClID because
On Sunday 26 December 2004 16:21, Steve Beaumont wrote:
Marc,
I'm not sure what you mean. Are you suggesting that I enter a sip entry,
E.g. [0870xyzabc] for the telappliant provided PSTN number ?
Yes.
I had to do something similar for my PSTN numbers over IAX - not near my * box
at present,
On Mon, Jan 03, 2005 at 07:11:46PM -0500, Garrett Smith wrote:
I wouldn't consider it an advertisement. There was no price, etc. I was
simply telling those that ordered from me previously I have more. It is
easier to send one email to 100's, then 100's of emails. If you do not want
to save
On Tue, 2005-01-04 at 15:20 +0200, Mark Elkins wrote:
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
phone number on my phone. In the log is the following - which displayed
'601' on my phone. The
Yes yes, your right. I forget these switches are smart!!! ;-)
-Original Message-
From: Julio Arruda [mailto:[EMAIL PROTECTED]
Sent: Monday, January 03, 2005 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] QOS / Cisco / Asterisk
Matt
Hello,
I had installed
several asterisk, but I every time had a problem with
callerID.
On each phones I
don't reveive the first digit.
For
example:
Caller 0672083516
called an IP Phone 0123456789. The IP Phone see 672083516 as
callerID.
I think there is a
patch for it, but I don't
On Mon, 2005-01-03 at 13:53 -0600, Matt Schulte wrote:
We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What
we're trying to avoid is hardcoding the IP address in the ACL. We were
trying to match by TOS set by Asterisk however it seems we've run into a
snag where the packet
Hello,
How can I access callers number with Asterisk CVS
1.0.12?
In new version there are structure cid with field
cid_num. And in 1.0.12 only callerid field which is equal to cid_name.
I also tried to get it from chan-cdr-src but
this is also the same as cid_name or callerid.
Hi all,
DIAX 0.9.9f is available for download (including the updated help file and
web page) from the following locations:
http://www.laser.com/dante
or
http://www.geocities.com/tdanro
You can see there what's new in 0.9.9f and the online help (for the
newcomers).
Please send me your feedback.
Just remember that CF cards have a limited life as far as read/write cycles
are concerned. Transcend says that their CF can handle 1,000,000
program/erase cycles before failure. I think, in practical experience, that
this is way high. My experience is between 50K and 200K.
---
Kelly D Griffin
First class service, thanks a lot :)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mattf
Sent: mardi 4 janvier 2005 03:48
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Manager API
There really isn't a solid
On Tue, 2005-01-04 at 15:45 +0200, Mark Elkins wrote:
On Tue, 2005-01-04 at 15:20 +0200, Mark Elkins wrote:
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
Playing with myself again - that is - I
i work myself on developing such an embedded asterisk. there is no major
problem, as long as you are careful when stripping out the underwood of the
core system.
my system boots of flash, loads /etc /var into ramdisk, so minimal write
cycles are wasted for configuration changes voicemail
Mark Elkins wrote:
On Tue, 2005-01-04 at 15:45 +0200, Mark Elkins wrote:
On Tue, 2005-01-04 at 15:20 +0200, Mark Elkins wrote:
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
Playing with
Adi Linden wrote:
I am experimenting with oh323 channels and h.323 gateways and a Cisco
CallManager. I am not using a gatekeeper at this time. Is it possible to
place calls coming into Asterisk from specific peers into specific
contexts?
Yes. You can associate called numbers/prefixes with contexts
Hi,
Is it possible to connect * with Avaya IP Office (central unit is IP412)?
Best regards
Micha
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Answer found :
Answer first, Ring, Wait, then ...
exten = s,1,Answer
exten = s,2,Ringing
exten = s,3,Wait(x)
exten = s,4,...
At 14:35 22/12/2004 +, you wrote:
Hello,
We have a DID partner sending traffic to Asterisk via SIP, but we are not
hearing ringtones. When we call the same extension
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steven Critchfield
Sent: Monday, January 03, 2005 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Qs about FXO/FXS cards
[...]
For business use, I
Hello,
This may interest you :
Place
termcapinfo xterm ti@:te@
in your .screenrc file.
(better yet in /etc/screenrc)
a b d o u l
aba at gcomnetworks.com
SIP: (131) 229-1002 at sip.freeipcall.com
___
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OK. You need to understand that retrieve_sip_conf_from_mysql.pl is a Perl
script and needs to be run outside of the sip.conf file.
Edit the script and put back the -T. Also edit the script to save to a
different file. Change it to sip_additional.conf.
Now run the script. It will connect to your
On Tue, Jan 04, 2005 at 08:05:16AM +0100, Florian Overkamp wrote:
On Mon, Jan 03, 2005 at 07:10:59PM +0100, Florian Overkamp wrote:
On Mon, 2005-01-03 at 17:38, Michael George wrote:
How are you transferring the call? With channel
facilities (e.g. hook flash)
or with the * '#'
Hi,
I just got
some interesting information from Kirk Telecom (www.kirktelecom.com). This company has
been in the business of providing DECT solutions (IP gateway, base stations,
repeaters and handsets)either to be used with Cisco CallManager (SCCP
protocol) or with the Innovaphone IP PBX
On Tue, 4 Jan 2005, abdoul wrote:
Answer found :
Answer first, Ring, Wait, then ...
exten = s,1,Answer
exten = s,2,Ringing
exten = s,3,Wait(x)
exten = s,4,...
That will cause all callers to you to have to pay even if the call is not
answered. Not very nice.
You may need to play with
[EMAIL PROTECTED] wrote:
What happens on the phone is that I hear voiptalk.org's greeting
and after they hang up, I hear my own invalid extensions
message.
That's because your _X. extension has a 2nd line. Reduce it to
_X.,1,Playback(invalid) and you won't have that problem.
Hi,
Is there some parameter that I should pay attention
to when using externip parameter on sip.conf?
I ask this because after using sipsak I've noticed
maybe the reason that i don't get voice in one direction could be because at the
SDP/SIP messages are references to the 192.168.1.50
What type of equipment does your DID provider have?
I had the same problem with Cisco and solved it by adding progress_ind
setup enable 3 on the voip peer.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson
Sent: Tuesday, January 04, 2005
Samudra E. Haque wrote:
Hello, anytime I make an IAX2 call to another peer, I am collecting CDR records
which are divided into small files, one for each accountholder customer that
makes the calls.
I have records of this nature:
123456,1234567890,IAX2/[EMAIL PROTECTED]/5,2004-12-30
GIBERT Frédéric wrote:
Hello,
I had installed several asterisk, but I every time had a problem with
callerID.
On each phones I don't reveive the first digit.
For example:
Caller 0672083516 called an IP Phone 0123456789. The IP Phone see 672083516
as callerID.
I think there is a patch for it, but I
What happens on the phone is that I hear voiptalk.org's greeting
and after they hang up, I hear my own invalid extensions
message.
That's because your _X. extension has a 2nd line. Reduce it to
_X.,1,Playback(invalid) and you won't have that problem.
I'm not the
First, please forgive me if this is a total newbie question, I've only
just begun to scratch the surface of asterisk.
I currently have a dialplan set up to let me dial a specific
extension, authenticate the user, then have * dial a hard
coded/programmed overseas number. What I would like to do
I have Obsequium running and have developed a way to parse the .PLS files
that it returns. Is there a way in Asterisk to play rtp:// streams as MOH?
Thanks.
--
Brian Wilkins
Software Engineer
[EMAIL PROTECTED]
Heritage Communications Corporation
Melbourne, FL USA 32935
Dear All,
we have installed a TE410P card on a Dell Poweredge 1750 running a slackware
10 with 2.4.26 Kernel. Then we have made two loops on the card and we have
configurated all the 120 channels. Our goals was to perform some stess tests
even if in this scenario we used the same box as generator
Steve:
I've problems with the new stack with outgoing calls and I need to revert to
the 0.0.1d. Under heavy load, some times one outgoing channel appears as
idle, but when I try to make a call, it returns a busy status, and keeps
the idle state, so the next try also is made in the same
On Tue, 4 Jan 2005, Justin Richards wrote:
I currently have a dialplan set up to let me dial a specific
extension, authenticate the user, then have * dial a hard
coded/programmed overseas number. What I would like to do is set up
my dialplan to have an extension that offers up an outbound
On Tue, 4 Jan 2005 10:08:27 -0500, Daryl G. Jurbala wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steven Critchfield
Sent: Monday, January 03, 2005 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Tue, 4 Jan 2005, Eric Wieling aka ManxPower wrote:
GIBERT Frédéric wrote:
Hello,
I had installed several asterisk, but I every time had a problem with
callerID.
On each phones I don't reveive the first digit.
For example:
Caller 0672083516 called an IP Phone 0123456789.
Does anyone have a reference (link?) with reviews of the most popular SIP
phones/ATA's can be found? We are looking to certify 3 or 4 VOIP phones
and/or ATA devices for use with * and need to purchase one of each to test.
If not, then what are the groups recommendations? The target customers are
Hi,
-Original Message-
Hmm, while this doesn't solve my problem it does point me
in the right
direction. I've just learnt that this is fixable behaviour
and it should
be/will change in cvs. I'll see what I can find out on that.
That would be great if they change it!
With the deafening silence from my previous questions, I feel seriously
alone in the desire to have ChanSpy available.
I want to be able to perform a ZapBarge on an Agents conversation, and
ChanSpy was the answer to my prayers.
Bug #2379 (http://bugs.digium.com/bug_view_page.php?bug_id=0002379)
snip
We're trying to PQ (Priority Queue)
packets on a Cisco using ACL's. What
we're trying to avoid is hardcoding the IP address in the ACL. We
were
trying to match by TOS set by Asterisk however it seems we've run
into a
snag where the packet TOS tends to get reset somewhere on our network.
Hi Roberto-
About a year ago, I ran extensive loopback tests of the kind you
described. I used various processors and motherboards, and used Fedora
Core 1 with 2.4 kernels. (See the asterisk-users message archives and
the wiki for more info).
I got similar results to yours, except that I had
Nice! that is exactly what I want!! much cleaner than the
callthrough plan (that I found two minutes after i posted my question)
I was trying to use that is posted on voip-info.org
Thanks!!
On Tue, 4 Jan 2005 17:24:36 +0100 (CET), Peter Svensson
[EMAIL PROTECTED] wrote:
On Tue, 4 Jan 2005,
At 03:25 PM 1/3/05 -0500, you wrote:
Unfortunately that makes Asterisk installs for small businesses more
expensive
than necessary. At US$500 for a T100P and US$300ish for a channel bank (FXS
only, FXO is significantly more expensive!) plus your time and system for an
Asterisk install it raises
Hi,
Anyone used this service, any comments on reliability/support?
Thanks
John
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Adi Linden schrieb:
...
In iax.conf eaxh peer has a context in which I can specify the context an
inbound call will be placed in. I don't see anything like this in the
oh323.conf file or the oh323 documentation.
Hi,
there is a workaround what is doing this job in most cases:
Use as general
On Tue, 4 Jan 2005 10:42:50 -0600, Michael B. Murdock wrote:
Does anyone have a reference (link?) with reviews of the most popular SIP
phones/ATA's can be found? We are looking to certify 3 or 4 VOIP phones
and/or ATA devices for use with * and need to purchase one of each to test.
If not, then
I have been picking at Asterisk for about a week, and I think I'm close. I
was hoping for a little guidance to bring this on home.
I want to be able to make outgoing calls from my SJPhone clients using my
VoicePulse Connect account. I have the two requisite items from Voice Pulse,
but I've had no
On Tue, 4 Jan 2005 16:57:46 +, John Middleton
[EMAIL PROTECTED] wrote:
Anyone used this service, any comments on reliability/support?
Works well for me. A hiccup on initial config was corrected quickly.
Peter
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I had such problem on PRI and BRI lines. And I think but I need to verify,
also on analog lines.
For example, I had such configuration:
[EMAIL PROTECTED] asterisk]# more zapata.conf
[channels]
switchtype=euroisdn
pridialplan=unknown
signalling=pri_cpe
group = 1
channel = 1-15,17-31
Jordi,
E s c a u x - Jordi Nelissen wrote:
1. It seems they foresee a SIP version of their product in Q1 2005.
Does this mean they will release an image for their '600' system? I'm
currently trying to implement it in a meaningful way in an office
enviroment, six handsets, but neither H.323
Hi,
I need some information on DID and Callback. Please read-on:
Question on DID (User1 Calling User2 via normal Telephone line and sending
its CLI:
Connectivity is as below:
User1 ==PSTN== DigiumE1/Asterisk1 ==INTERNET== DigiumE1/Asterisk2
==PSTN== User2
1. Can User1 make a single stage call to
check this link:
http://www.iptel.org/info/products/sipphones.php
João Pereira
- Original Message -
From: Michael Graves [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 04, 2005 5:00 PM
Subject: Re:
Anyone know how to get app_queue to send logs to MySQL or
any other sql server.
I found info for cdr's and even configs but nothing on
queue_log.
If sql is not supported in the current app_queue I will be
willing to pay someone to add it.
John Bittner
Simlab.net
cmould wrote:
Hi:
Just saw your post while trying to solve a similar asterisk problem. Did
not see any responses. Was your problem solved and what was the solution?
Carey
To what problem are you referring? There are multiple issues I've
discussed on the list and I can't remember the topic of
Greg - Cirelle Enterprises wrote:
When you try to sell the asterisk system, you have to compete with that and
frankly, all the people want is to make phone calls.
Mention voice over ip and eyebrows raise, I've heard of that, but in reality
nobody cares how their phone calls are made, just that it
At 11:34 AM 1/4/05, you wrote:
On Tue, 4 Jan 2005 10:08:27 -0500, Daryl G. Jurbala wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steven Critchfield
Sent: Monday, January 03, 2005 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial
On Tue, 4 Jan 2005, John Middleton wrote:
Hi,
Anyone used this service, any comments on reliability/support?
Hi John,
I've been using this service for a while now, works very well.
Had a couple of minor problems with IAX originally, but they were sorted
within a couple of hours, and they even
On Tue, 4 Jan 2005, GIBERT Frédéric wrote:
I had such problem on PRI and BRI lines. And I think but I need to verify,
also on analog lines.
For example, I had such configuration:
[EMAIL PROTECTED] asterisk]# more zapata.conf
[channels]
switchtype=euroisdn
pridialplan=unknown
Michael Graves wrote:
That might work out where you do your deployments. In Verizon
territory, you can get analog business lines with unlimited long
distance and no metered minutes for about $37 a month. A BRI costs you
about double that for the loop, with metered minutes and bring your own
LD.
Sipura SPA 3000... forget the channel bank and PRI card. Buy a PRI card
and ebay the SPAs when you arte ready to move from POTS to PRI, or
better yet, forget both and find an ITSP that can offer QoS (private
line!!!) and interface with *
Talkswitch? Get on the VoIP bus or get run over buy it,
On Tue, 2005-01-04 at 10:08 -0500, Daryl G. Jurbala wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steven Critchfield
Sent: Monday, January 03, 2005 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Sorry list - this was meant to go off list.
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[EMAIL PROTECTED] wrote:
On Mon, 3 Jan 2005 [EMAIL PROTECTED] wrote:
I've been trying all day to reach their techie folks to ask a couple
questions about something I worked all weekend on. Keeps rolling to VM
and the receptionist does the same thing.
Just was wondering if anyone else was able to
The new configuration style of OH323 will simplify the sections of
the dialplan that handle H.323 calls.
Michael.
Roger Schreiter wrote:
Adi Linden schrieb:
...
In iax.conf eaxh peer has a context in which I can specify the
context an
inbound call will be placed in. I don't see anything
Customer is having problems with his internet connection, I
have in my context:
[jimballboutiques]
exten = 1235690251,1,SetGroup(customer)
exten = 1235690251,2,CheckGroup(3)
exten = 1235690251,3,Dial(SIP/jimball,20,r)
exten =
1235690251,4,VoiceMail([EMAIL PROTECTED])
exten =
That's an LD T1 PRI right? Long distance only? No tone?
What does a local T1/PRI cost? That gets unlimited intra-LATA calling?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg - Cirelle
Enterprises
Sent: Tuesday, January 04, 2005 12:29 PM
To:
Then in extensions.conf or better in a file like oh323peers.conf
included in extensions.conf switch to contexts per peer via gotoifs:
This will work ok for my purposes, where I work with CallManager and
gateways.
Thanks,
Adi
___
Asterisk-Users
[not-in-service]
exten = _.,1,Answer
exten = _.,2,Wait(1)
exten = _.,3,Playback(the-number-u-dialed)
exten = _.,4,Playback(not-yet-assigned)
exten = _.,5,Wait(2)
exten = _.,6,Hangup
The above is included as the last of several includes, forcing it to
be the last choice if no other
You mean can User 1 just pickup the phone and call User2 with just dialing 1
number? Seems easy enough, In asterisk1:
exten = 123456789,1,Dial(IAX2/[EMAIL PROTECTED]/123456789)
Then in Asterisk2:
exten = 123456789,1,Dial(ZAP/g1/987654321)
Now whenever somebody calls 123456789 and it goes into
Hi,
I am experiencing one-way audio from:
SIP Device Asterisk - Cisco 7200
The Cisco 7200 has a VXE+ card that will allow you to do SIP. I can pass
audio from SIP Device to Asterisk through the Cisco 7200 to the other end,
but the Cisco 7200 does not return any audio back to
Hi John,
I've been using this service for a while now, works very well.
Had a couple of minor problems with IAX originally, but they were sorted
within a couple of hours, and they even called me to tell me they'd fixed
it.
If you use SIP with them though, you may find weirdness with incoming
Anybody know anything about this F-1000 phone?
100 hours of battery life, not bad at all...
http://www.lightreading.com/document.asp?site=lightreadingdoc_id=65310
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
What's wrong with doing it by port?
We're actually using SIP to terminate calls, going by rtp.conf the ports
could range several thousand ports. What we're going for is only
honoring TOS for that particular customer, luckily these are T1
customers hosted on our routers. They understand that
Andrei (MPI) wrote:
Michael Graves wrote:
That might work out where you do your deployments. In Verizon
territory, you can get analog business lines with unlimited long
distance and no metered minutes for about $37 a month. A BRI costs you
about double that for the loop, with metered minutes and
The problem is solved.
With the AVM-Fritz Driver everything is working.
Current working setup:
Fritz!PCI - Proprietary-AVM-Fritz-Driver(http://avm.de) - chan_capi -
Asterisk
Non working setup (no dialtone):
Fritz!PCI - mISDN - chan_capi - Asterisk
regards
Stefan
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