I am trying to route my calls through an outside IAX provider. I
am having a problem with which codec to use. The only way I have
successfully been able to make an outgoing call is if i do:
disallow=all
allow=g729
in the sip.conf file (for my phones) and the iax.conf file. The second I add
On Wed, November 23, 2005 19:21, Jose Limeres said:
I also had good results with mISDN where CAPI and BRIstuff failed
before. In my case just for 1 HFCPCI BRI card.
The instalation ran smoothly except that in the end when I try to
restart ASterisk get the message: Unable to initialize mISDN.
I need help connect Cisco router M3800 serries to asterisk
this is my configuration has anyone done this I have done everything on the
http://www.voip-info.org/wiki-Asterisk+cisco+FXO
but I dont get any dial tone. Please help
voice-port 1/1
input gain 10
output attenuation 10
no
I'm working on a manager client that I designed to hold open TCP
connection to asterisk while it is running for varoius purposes. After
being puzzled by unexpected behavior, I realized that the server closes
the connection after it completes an originate action - or at least it
does in the
On Wed, November 23, 2005 11:17, Francesco Peeters said:
On Wed, November 23, 2005 7:21, Tzafrir Cohen said:
On Tue, Nov 22, 2005 at 09:20:23PM +0100, Francesco Peeters wrote:
I have seen several claims that it can be done: Multiple HFC-PCI cards
running both BRI_CPE_PTMP and BRI_NET_PTMP
Hi list:
what does this message mean Warning LSP Low and does
it relate to the bandwidth?
__
Yahoo! Mail - PC Magazine Editors' Choice 2005
http://mail.yahoo.com
___
--Bandwidth and Colocation
Title: Aastra 1.3 firmware
On Wed, 2005-11-23 at 11:08 +, Lee Archer wrote:
Has anyone had any luck with the BLF option yet? I have set up as per the manual/front end, configured the hints in Asterisk and nothing shows.
Regards
Lee
It works for me but if you
NON-DISCLOSURE: ALL CUSTOMERS USING VOIPJET'S SERVICE ARE SPECIFICALLY
PROHIBITED FROM DISCLOSING TO OTHERS THAT THEY USE VOIPJET'S SERVICE,
I use Voipjet,
I have used Voipjet...
Did I mention I use Voipjet?
I'd like to teach the world to sing (about using Voipjet)...
So sue me Voipjet,
I am using a Cisco 1760V with FXO card in Australia to provide ports into
Asterisk.
I was wondering if anyone out there has a config for the cisco to detect
the disconnect or hangup signal for Australian tones.
If the calling party hangs up while leaving a voice mail for example, it
takes
On 11/23/05, Bill Michaelson [EMAIL PROTECTED] wrote:
I'm working on a manager client that I designed to hold open TCP
connection to asterisk while it is running for varoius purposes. After
being puzzled by unexpected behavior, I realized that the server closes
the connection after it
Far as I know these products are going to be tied to the LinksysOne
hosted services program, which you can find more information on at
www.LinksysOne.com
Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY 14225
hello
___
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Téléchargez cette version sur http://fr.messenger.yahoo.com
On Wed, November 23, 2005 20:47, Chris Mason (Lists) said:
NON-DISCLOSURE: ALL CUSTOMERS USING VOIPJET'S SERVICE ARE SPECIFICALLY
PROHIBITED FROM DISCLOSING TO OTHERS THAT THEY USE VOIPJET'S SERVICE,
I use Voipjet,
I have used Voipjet...
Did I mention I use Voipjet?
I'd like to teach the
Hello Everyone!
For all of you PhoneCALL users, we have a treat for you today as
PhoneCALL 2.7-RC1 has been released!
We've worked hard to make this release as close to as bug-free as
possible, but in the event you find a bug - PLEASE report it to the
bugtracker. It doesn't matter how small
hello
___
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Téléchargez cette version sur http://fr.messenger.yahoo.com
On Nov 23, 2005, at 11:14 AM, Michael wrote:
I am trying to route my calls through an outside IAX provider. I am
having a problem with which codec to use. The only way I have
successfully been able to make an outgoing call is if i do:
disallow=all
allow=g729
in the sip.conf file
David Waugh wrote:
Yes, you can use the Eicon Diva Range with 2.6 Kernels
Another question, considering the card should arrive tomorrow and I'd
like to try my hand at setting it up this weekend: Do I need to BRIstuff
Asterisk to get the Eicon Diva V-4BRI to work, or should I just need
Hi Dave,
exten = callpark,1,Dial(SIP/1000) didn't work - invalid extension
exten = callpark,1,Transfer(1000) didn't work - the parker hung up,
and the stall number announcement was made to the parked caller.
On Nov 22, 2005, at 10:34 PM, David Hindmarsh wrote:
Hi Guys,
What happened if
Others may know better than me, but I don't think so...
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Nov 22, 2005, at 11:02 PM, Marcus Deluigi ((intern)) wrote:
That helped a little.
Yes?
harry gaillac wrote:
hello
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Hi all
I'm having trouble receiving faxes using rxfax. Could somebody please browse
my log file and give me a swift kick in the right direction? I've also added
my zapata.conf file at the end.
I tried adjusting the rxgain to 15 (as mentioned in the archives) but that
didn't seem to make a
On Thu, 24 Nov 2005, Avi Miller wrote:
David Waugh wrote:
Yes, you can use the Eicon Diva Range with 2.6 Kernels
Another question, considering the card should arrive tomorrow and I'd like to
try my hand at setting it up this weekend: Do I need to BRIstuff Asterisk to
get the Eicon Diva
On 11/22/05, Marcus Deluigi (intern) [EMAIL PROTECTED] wrote:
That helped a little.
Thanks a lot!
Is there any chance to determine the agent id (defined in agents.conf)
of a caller?
If I'm understanding you correctly, you seem to be under the
impression that you can only use
Title: Message
Hi,
I'm trying to use
modems with Asterisk+VoIPGatewaysin an attempt atproviding
anInternet service.
Home_PC--Modem--PSTN--VoIP_Gateway_FXO--Ethernet--Asterisk--Ethernet--VoIP_Gateway_FXS--Modem--PPP_Server--Internet
I've been trying to
use G711u and G711a codecs on the
This is only slightly related to Asterisk (in that we are using
Asterisk as our PBX),
so feel free to contact me off-list.
I need to hear from someone who has practical experience using the combo of
Asterisk, WiFi, and WiFi VOIP phones like the Hitachi-Cable IPC-5000.
In particular, I need to
Well I am using a wifi phone running sip and works fine. Using commercial
grade access point. With Prizim card.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Tillman
Sent: Wednesday, November 23, 2005 1:34 PM
To: Asterisk Users Mailing List -
On Wed, 23 Nov 2005 15:33:49 -0600, David Tillman wrote:
This is only slightly related to Asterisk (in that we are using
Asterisk as our PBX),
so feel free to contact me off-list.
I need to hear from someone who has practical experience using the combo of
Asterisk, WiFi, and WiFi VOIP phones
Has anyone has any luck setting up cisco MC3800 with asterisk. I looked at
the example on the voip-info.org site, but know luck. Someone pls help
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Depending on load, and this shouldnt be a problem with low call volume,
wifi is half duplex, so you have to take that into account.
I however use a ipaq h5500 with a softphone running and it works fine
over wifi, while that isnt quite the equipment you requested info on, I
do not have any
On 11/23/05, Michael Graves [EMAIL PROTECTED] wrote:
I eventually switched to using a Astra 480i CT desk phone with a couple
of corless handsets. It's been great.
My first thought was to use a cordless phone and a Sipura ATA. But this
is a 100,000 sqft warehouse with a freezer section in the
On Wed, 23 Nov 2005 15:57:25 -0600, David Tillman wrote:
On 11/23/05, Michael Graves [EMAIL PROTECTED] wrote:
I eventually switched to using a Astra 480i CT desk phone with a couple
of corless handsets. It's been great.
My first thought was to use a cordless phone and a Sipura ATA. But this
Hi Matt,
I did not move the whole asterisk directory I just put a link to it. (ln -s
/usr/src/asterisk-1.2.0 /usr/src/asterisk)
Then I tried to compile but the error stayed.
I also tried with MySQL 4.1.15 and had the same error.
I am getting to the point where I think I might have not all
I concur with Michael, the current crop of WIFI phones on the market do
have their individual quirks, and you will likely encounter issues using
consumer grade access points. If you have some money to throw at this,
and want a real slick, industrial grade solution that will integrate
with
snacktime wrote:
On 11/23/05, Bill Michaelson [EMAIL PROTECTED] wrote:
I'm working on a manager client that I designed to hold open TCP
connection to asterisk while it is running for varoius purposes. After
being puzzled by unexpected behavior, I realized that the server closes
the
Does Asterisk 1.2 support INVITE with Replaces header (rfc 3891) ?
thanks
- Arnaud
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You can try Zyxel Wifi phones
They do the JOB cost bout 200 bucks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews
Sent: Wednesday, November 23, 2005 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I think that timing issues will kill you, but if it were going to work
you would want to use ulaw all the way around as your codec.
a better option would be to use more traditional terminal server/remote
access server type hardware off of an actual copper pstn line.
you can pick up terminal
Does asterisk fully support DNS SRV lookups yet, or does it still only
read the first SRV entry?
Info on the wiki looked quite old, so I thought I better ask.
regards
David
___
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Asterisk-Users
FWIW, the cordless handsets on the Aastra 480i CT work well. I've waled
about while on a call, to a distahce of approx 300 feet before the call
quality started to suffer. The phone supports up to 8 (I think)
handsets, treating each as a separate extension. It can page between
handsets without
Yeah the 480i-CT is a nice product, and not subject to the
inconsistencies of WIFI. What impressed me about the Engenius handsets
was the range. 250,000 square feet coverage, or 3,000 acre coverage in
an outdoor application, off a single radio AP, that's a large footprint.
Cory J Andrews
Thank you to everyone for the input. It may be to our advantage to
install a Wi-Fi mesh
with handoff as we will eventually put data-terminals on our
fork-trucks. In the meantime,
we have warehouse managers carrying cell phones for comms to the office.
I'm going to drop five Snom or Grandstream
On Wed, 2005-11-23 at 19:58 +0100, Fred Blaise wrote:
Hi all
I am having an issue when trying to 'make astman'. Asterisk 1.2.0 here,
from source, on debian sarge. Everything else working fine (only SIP
setup anyway)
deafneuron:/opt/asterisk-1.2.0/utils# make astman
cc -DNO_AST_MM -c -o
Wolfgang S. Rupprecht writes:
If there is enough interest, maybe the greater asterisk community
could adopt some semi-official mapping tables. I'd be willing to
periodically generate a flat mapping file and an extension.conf
dialplan snippet from sipbroker's list or whatever else is deemed
Yes it would be really interesting if there are any IAX libraries for
Java that are available under an open source license and that we might
improve further.
There is a growing demand for such a thing (for example see
http://forums.digium.com/viewtopic.php?t=2431)
Would be cool if we can create
David - the 2nd generation crop of WLAN handsets will start coming to
market shortly, and vendors are promising improvements. UTStarCom has
the F3000 coming in December, which will have according to their spec
* WEP (64 and 128 bit )/WPA/MD5 Auth
* Handover/Roaming between different AP
On 11/23/05, Bill Michaelson [EMAIL PROTECTED] wrote:
snacktime wrote:
On 11/23/05, Bill Michaelson [EMAIL PROTECTED] wrote:
I'm working on a manager client that I designed to hold open TCP
connection to asterisk while it is running for varoius purposes. After
being puzzled by
Justin,
I can tell you that I haven't been able to get a clearwire rep out to my
location to demonstrate their lines to us. I keep calling, telling them
what i want to do. And they tell me that clearwire is a great service,
and that one of they will relay my questions to a sales rep who
Klaus Darilion [EMAIL PROTECTED] writes:
There is a new ietf WG to come which deals with peering issues. It's
called SPEER (formerly VOIPEER)
The list archive is at
http://darkwing.uoregon.edu/~llynch/voipeer/
minutes from last ietf meeting:
Stefan Reuter wrote:
Yes it would be really interesting if there are any IAX libraries for
Java that are available under an open source license and that we might
improve further.
There is a growing demand for such a thing (for example see
http://forums.digium.com/viewtopic.php?t=2431)
Would be
On Wed, 2005-11-23 at 16:29 -0700, Jason Becker wrote:
http://www.hem.za.org/jiaxclient/
Thanks for the pointer.
I should have been more clear with my request: What I am looking for is
a pure Java implementation. JIAXClient is a solution that is ok for many
use cases but is unacceptable in
I'm using Cisco 7960 phones with asterisk. When I dial extension 200 from
my phone, it displays on the screen that I'm dialing 200. Is there a way
to have the phone look up the callerid value in sip.conf and use that
information instead of the dialed extension number?
Hi,
I have one Asterisk linked to a MD110 (Ericsson PBX) using a TE100P. I'm
using the QSIG ( Asterisk 1.2).
From * I can make calls elsewhere. But when the calling is coming from
MD, the Asterisk is answering the call at the first digit it receives.
The dial plain is waiting for a four
I don't know of any way of doing it thru asterisk, but you can test to
see if 200 exists in the 7960 directory xml file, maybe it will take
the value from there, but I might be wrong. I don't have one in front
of me to test it.
On 11/23/05, Jeremy Koski [EMAIL PROTECTED] wrote:
I'm using Cisco
Hi all,
Anyone know if the following message is something that should not
happen? I'm running asterisk as user/group asterisk/asterisk.
-- Executing VoiceMailMain(SIP/1003-6e02, s1003) in new stack
Unable to create lock file
'/var/spool/asterisk/voicemail/default/1003/Old': No such file or
That helped a little.
Thanks a lot!
Is there any chance to determine the agent id (defined in
agents.conf)
of a caller?
If I'm understanding you correctly, you seem to be under the
impression that you can only use
RemoveQueueMember/AddQueueMember on agents that are defined
Hi, All
Does any one has successful experience use te410p and spandsp together?
Could they work well with all 120 channels receive/send fax at the same time?
My practice is that rxfax always get broken fax page.
Help!___
--Bandwidth and
On Wed, 2005-11-23 at 15:18 -0800, Wolfgang S. Rupprecht wrote:
Wolfgang S. Rupprecht writes:
If there is enough interest, maybe the greater asterisk community
could adopt some semi-official mapping tables. I'd be willing to
periodically generate a flat mapping file and an extension.conf
Is there a solution for the problem that the card in use flag is set,
after the user hang up?
The flag remains set, if the user hang up, after the price for the call
will be announced.
It is bad (for the business), because this happens most of the time only
for NEW users!
Solutions?
1. Do
Ronald Wiplinger wrote:
Is there a solution for the problem that the card in use flag is set,
after the user hang up?
Yes, there is a patch. This was fixed in cvs quite a while ago.
Put this:
$SIG{HUP} = 'ignore_hup';
sub ignore_hup {
print STDERR \nHUP received!\n\n;
}
just after
Hi,
I have a very strange Asterisk SIP call signalling problem that is proving
extremely difficult to track down. The problem is that any SIP INVITE
request that is coming into Asterisk over a satellite connection from a
Linksys Router or PAP2 is getting a Not Acceptable Here (codec error) from
If you use qualify=yes to determine whether that device is alive or not,
then it won't be very accurate as every now and then, the device may
fail to reply to the SIP OPTIONS packet due to reasons other than it is
really offline.
If you are linked to a PSTN GW, I would believe that GW will
Patrick [EMAIL PROTECTED] writes:
Shouldn't the last line in exten-peers.conf be:
exten = _**XXX.,1,Macro(dial-outgoing,SIP/${EXTEN:[EMAIL PROTECTED])
^^^
Similar to the previous line sipbroker line:
exten =
I brought this up a while back and althought there are pieces that
interface * into Fax Telephony applications, there hasn't been something
that works with plain old analog modems.
Then I found this piece of code. From my initial tests it looks solid,
but I have no clue in how to interface this
Hi:
Well I tried to connect to instances of asterisk (hylafax iaxmodem),
but I have a problem. I tried to send a fax from one of them, a message
appears:
-- Registered IAX2 '4000' (AUTHENTICATED) at 127.0.0.1:33270
-- Accepting AUTHENTICATED call from 127.0.0.1:
requested format =
Whoops... Sorry.. Mailer delay.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don
Fanning
Sent: Wednesday, November 23, 2005 5:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Virtual Modems Revisited
I brought this up a while back
Title: Message
Are
you also implementing the "ping" keepalive as part of your
app?
Mark
-Original Message-From: Bill Michaelson
[mailto:[EMAIL PROTECTED] Sent: Thursday, 24 November 2005 9:09
AMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Re: manager
Don Fanning wrote:
I brought this up a while back and althought there are pieces that
interface * into Fax Telephony applications, there hasn't been something
that works with plain old analog modems.
Then I found this piece of code. From my initial tests it looks solid,
but I have no clue in
Jeremy Koski wrote:
I'm using Cisco 7960 phones with asterisk. When I dial extension 200
from my phone, it displays on the screen that I'm dialing 200. Is there
a way to have the phone look up the callerid value in sip.conf and use
that information instead of the dialed extension number?
At
Kevin P. Fleming wrote:
Matt Riddell wrote:
So how does Asterisk know that the media stream has been disconnected
between
the two remote hosts?
It doesn't... nor does any other SIP softswitch. See my other reply for
a possible solution.
I agree that you could code a fix, but saying my
Aaron Clauson wrote:
m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
I don't know
I had the same problem when we were setting up these boxes after
katrina. What i found is that they will only do one G729 session at a
time. so that mesg that your showing is that its trying to
register two chans as 729. what i did to get around this was to
turn off fource prefered codec on one
Matt Riddell wrote:
The fact remains, if you need *very* accurate cdr's then you either don't do
canreinvite=yes for the peer or you code something so that Asterisk notices
that the rtp has stopped. The fact remains that without these, the most
accurate CDR is going to come from the provider.
Trust me this is on the ATA. set both lines to use 729 but dont fource
them to only use that codec (in the ata config) I spent days trying to
figure this out the first time i ran accross it, and after that config
change on the ata i haven't had problems. I have seen this on most of
the sipura's
Hi Roger,
We've solved this with the MD110 sending calls to cisco VoIP gateways.
The method is to set the Minimum and Maximum call length for this number
range on the MD110 - and to configure the destination route to only send
the call when the minimum length is reached (sometimes called
Greetings,
Does anyone know of aAsterisk Manager
Interface client application that can run from a Windows XP machine to manage
Asterisk installed on a Linux Machine.
I just know something like this exists but can't
seem to find it out there.
Thanks,
KC
Hi,
Thanks for the tip I'll try it out. That would explain some situations where
one of the peeople concerned was mucking around with the codec settings on
the PAP2 and managed to get some calls out.
It's a bit baffling how the Linksys devices will get INVITES through without
G.729 being set
I have fedora core 2 on this box.
I updated to the latest kernel but same problem.
My kernel is 2.6.10-1.771_FC2
aNY IDEAS
bART
I had this problem with Fedora. I updated the kernel to the latest one
available for core 3 and changes the links to point to the new source
code. It worked fine
check out http://ipswitchboard.thorben.dk/ there is an asterisk
manager and other nice GUI 's for Windows
On 11/23/05, kchase [EMAIL PROTECTED] wrote:
Greetings,
Does anyone know of a Asterisk Manager Interface client application that can
run from a Windows XP machine to manage Asterisk
Leo Burd wrote:
Hello everyone,
I'm implementing an audioblog application and have some questions about
how to best stream and/or convert MP3 and WAV files to be played by
Asterisk. Currently, I first copy the files from the server to my
machine, convert them to Wav and play.
Bartosz Wegrzyn - asterisk wrote:
I have fedora core 2 on this box.
I updated to the latest kernel but same problem.
My kernel is 2.6.10-1.771_FC2
I had this problem on Ubuntu 5.04 when I checked out the 1-2 and 1-2-0
cvs tags, just downloading the tar.gz from the asterisk site worked
fine,
Title: Codec negotiation (not the same old stuff)
I have a H.323 device, let's call it stupid, that supports all variants of G.729. That should be
good, but no. When it negotiates a call between Asterisk and a phone that supports all
varients of G.729, it gets it wrong. Asterisk sends
Hello
Sorry to tell you that I am resending this mail because didn't get a
reply for this query.
Salil
Hello
I have seen the article in digium site about the answering machine made
using a softmodem and the zap library. I am using Fedora Core 2/3 system for
doing this
Thanks it helped
Bartosz Wegrzyn - asterisk wrote:
I have fedora core 2 on this box.
I updated to the latest kernel but same problem.
My kernel is 2.6.10-1.771_FC2
I had this problem on Ubuntu 5.04 when I checked out the 1-2 and 1-2-0
cvs tags, just downloading the tar.gz from the asterisk
Hello everybody :-)
This are my first line french zapata.conf settings.
I have 3 like this, with only rx/tx gain a little bit different levels.
Running well.
Best Regards,
Francois BERGERET,
France.
usecallerid=yes
hidecallerid=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
El lun, 19-09-2005 a las 14:28 +1000, Shaun Ewing escribió:
Some of you may remember back in May the thread on using Asterisk as a
voicemail server for a Cisco Callmanager system.
I'm looking for the same feature, please let me know where can I find
more resources and info about it.
Thank
harry gaillac wrote:
hello
Hi there! How are you today?
--
Cheers,
Matt Riddell
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Kevin P. Fleming wrote:
However, in the general case of not being concerned so much about the
peer going away and losing CDR information for _one_ call, using
reinvites does _not_ impact the quality of the softswitch's (Asterisk)
CDRs.
Agreed.
--
Cheers,
Matt Riddell
Dear All,
Can I use Asterisk IP-PBX as Softswitch? If not, what
is lacking in asterisk
from not *becoming* softswitch?
Thanks
Regards,
Somesh S. Shanbhag
__
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Ah... Well I was sort of thinking more along the lines of trying to get
this to work into IAX or SIP. But if you know for sure that the
modulation is broken...
Just imagine... You'd be able to have a modem bank and save thousands of
dollars in leasing/purchasing a modem bank.
-Original
I would go with chan-capi-cm, as well as loading up the eicon drivers
first for the base drivers and utility set.
I have a few installations as such that are working flawlessly, and
Armin has done great work on the driver.
Regards,
Greg
-Original Message-
From: [EMAIL PROTECTED]
On Nov 24, 2005, at 12:06 AM, Scott Clements wrote:
HI List,
You'll have to pardon the newbieness of this question, I was
editing the sip.conf file on my asterisk server yesterday, and now
none of my asterisk trunks will connect. From my knowledge sip.conf
does not effect registration,
harry gaillac wrote:
hello
Hi there! How are you today?
Very well, thank you.
PaulH
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Okay, finally run a server up with asterisk 1.2 and started work on getting CCM 4.1 talking to it to try and investigate the use of * for voicemail and possibly meetme conferences. Using the notes on the voip wiki, I've managed to get it to a point where I can call * successfully and get into and
hi, i was able to do a make linux26 without problem on my FC2 machine.
but when i tried a make install nothing has been install.
i had another machine running FC4, doens't have this problem. any ideas?
thank you.
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Hi,
I'm now using Asterisk for my voicemail together with SER.
They just work fine. When the user in SER is not registered the
call will be forwarded to Asterisk and the caller will record his
message. Then I also made asterisk to send the wav as attachment to
its email. I try
Check the location specified in the kernel Makefile, and validate that is
installs the modules to the propler /usr/lib/modules/bla blabla directory.
Nir S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kong
Sent: Thursday, November 24, 2005 9:01 AM
To:
Kevin P. Fleming wrote:
Matt Riddell wrote:
So how does Asterisk know that the media stream has been disconnected
between
the two remote hosts?
It doesn't... nor does any other SIP softswitch. See my other reply for
a possible solution.
...or implement the SIP timer extension.
/O
Kevin P. Fleming wrote:
David Thomas wrote:
Is the CDR accounting done based on SIP signaling? If a UA is talking
(RTP) to a third party PSTN gateway, isn't it at risk if say the UA
loses power. How will asterisk know the call has ended if it is not
involved in the media path. The idea is
Tzafrir Cohen wrote:
On Tue, Nov 22, 2005 at 02:00:11PM -0700, Matt wrote:
Looks like you need to install the kernel headers package. While you
are at it be sure that you have the kernel source package installed also.
apt-get install kernel-headers-`uname -r`
should suffice.
old zaptel 1.0 install nicely, how come the new one will have problems?
At 03:09 PM 11/24/2005, you wrote:
Check the location specified in the kernel Makefile, and validate that is
installs the modules to the propler /usr/lib/modules/bla blabla directory.
Nir S
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