Hi,
Any Idea about this message error are welcome, this cause communication
cuts.
== D-Channel on span 1 down
MFE for TEI = 84
== D-Channel on span 1 up
this is my zapata.conf file
;signalling = bri_cpe
pridialplan = unknown
;echocancel = yes
echocancel = no
immediate=no
; BRI line plugged into
I was wondering if anyone has setup an * connected to
E1 in Mexico?, what card would you recomend and do you have some info, examples
or everythig else... or for instance this setup works?
Thanks in advance
Greetings Otto Krumm
Hi,
Any Idea about this message error are welcome, this cause communication
cuts.
== D-Channel on span 1 down
MFE for TEI = 84
== D-Channel on span 1 up
Best Regards.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/l
Turned out to be a dumb mistake on my part.
when su'ing to my non-root user, I was losing my PATH info.
safe_asterisk assumes /usr/sbin is in the PATH.
Cheers
Ryan
On 31-Mar-04, at 5:59 PM, Ryan Courtnage wrote:
Hello,
I've found a couple of previous posts on this subject, but with no
posted
Aaron Martin wrote:
Has anyone heard any more info about the Virbiage FT201 VoIP phones?
About 3 months ago I was told they were 6 weeks away, about 3 weeks
ago I was told they were 2 weeks away, and now I am told they are 2
months away again! Are they EVER going to arrive? Can anyone shed
(BIs this same result?
(B
(Bhttp://lists.digium.com/pipermail/asterisk-users/2003-November/027321.html
(B
(Bmack_jpn
(B
(BOn Thu, 1 Apr 2004 01:08:44 +0900
$B4dED(B $B?-2p(B <[EMAIL PROTECTED]> wrote:
(B
(B> hi.
(B>
(B> I got these compile errors while install asterisk.
(B> readli
Has anyone heard any more info about the Virbiage
FT201 VoIP phones?
About 3 months ago I was told they were 6 weeks
away, about 3 weeks ago I was told they were 2 weeks away, and now I am told
they are 2 months away again! Are they EVER going to arrive? Can
anyone shed some light on thi
This is the version with the problem.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Wednesday, March 31, 2004 11:35 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Extension ringing but no ringing sound.
On Wed, 2004-03-31 at 2
On Wed, 2004-03-31 at 20:57, Gene Kochanowsky wrote:
> Thanks Eric. Is this the CVS branch you are referring to? -
>
> # cvs checkout -r v1-0_stable asterisk
Yes.
--
Useful Asterisk Docs:
http://www.digium.com/index.php?menu=documentation (look at the
"Unofficial Links") and http://www.voip
I am looking for a way to set my * so that it'll accept REGISTERs with a
particular username, regardless of the password used. Snooping the network
traffic suggests that the password is sent in digest or some other
non-plaintext format, so I don't suppose there is an easy way for me to
find out wha
The v1-0_stable version is the one with the problem. I have checked out
the latest and it does ring.
Gene
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gene
Kochanowsky
Sent: Wednesday, March 31, 2004 9:57 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk
> This error probably indicates something significant. What
(B> sort of system are you compiling on? It's quite unusual for
(B> "pwd" to be missing on any unix-like system. So it must not
(B> be in your search path. It's almost always /bin/pwd, and if
(B> you don't have /bin and /usr/bin
Could I do things like call other ext on the system? Check Voice mail? I
would like to test this before I put money in cards I may not need. What
Software Phone app is people using?
Thanks for all the help so far.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On B
Thanks Eric. Is this the CVS branch you are referring to? -
# cvs checkout -r v1-0_stable asterisk
Gene
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Wednesday, March 31, 2004 8:46 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asteris
Perhaps the 7960 has "Call Waiting" set to "YES" - this being the case, you're
really not hitting a "busy" but the 15 sec "timeout" instead.
Try setting "CallWaiting=No" in the 7960 you should then get a 'busy'.
Oh, another thing. If you have multiple line appearences configured as the same
SIP p
On Wed, 2004-03-31 at 19:05, Gene Kochanowsky wrote:
> Greetings,
>
> This is probably some configuration issue, but for some reason my system
> has stopped playing a ringing sound when an extension is dialed. The
> phone rings but there is no ring sound in the ear piece.
You updated your Asteris
Hi Mark,
I don't think it is codec mangling. My entire system is TDM. On incoming
calls, after asterisk picks up there is a ring but that is the only time
I hear it.
Gene
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday,
Hi Gene,
I had this problem. I found that it would work between
devices that shared the same codec settings but not
otherwise.
Try adding |T to the end of your extensions dialing
string. This will force * to handle the call and so do
any codec mangling. This has a hit on your server but
at least
This has been discussed before. Insert a "wait" in your dialplan before
the voicemail starts pumping out the rtp stream. It seems the Cisco
phones are slow to set up the audio stream.
Michael Welter wrote:
I'm having problems with the voicemail prompts. The beginning of each
prompt is garbl
Greetings,
This is probably some configuration issue, but for some reason my system
has stopped playing a ringing sound when an extension is dialed. The
phone rings but there is no ring sound in the ear piece.
Gene Kochanowsky
___
Asterisk-Users mailin
Hello,
I've found a couple of previous posts on this subject, but with no
posted resolution...
I'm attempting to run * as a non-root user (asterisk), following the
guidelines on the wiki:
http://voip-info.org/tiki-index.php?page=Asterisk%20non-root
I can run * as my new user with "/usr/sbin/
Hi Alex,
Alex Zarubin wrote:
Hi Steve and all,
1. Faxing from asterisk back to the same asterisk (from one Zap
channel to another)
doesn't work for us. Txfax called with the 'caller' parameter
issues CED, while the
receiving side needs CNG in order to switch to fax extension
w
Please send me a copy
On Thu, 1 Apr 2004, Leo Ann Boon wrote:
> Figured this might be interesting to some on the list. I've written a
> small tool to combine the 2 res_monitor created wav files into a single
> compressed (IMA ADPCM 8KB/s) stereo file. During playback, you can
> select the part
How do I do this
1) ZAP-> * -> IAX(1) --> IAX(2) -> DG104S --> Handset
2) No Answer on Handset
3) Back to IAX(1)
4) IAX(1) tries a cell phone
5) Still no Answer
6) Local * Voicemail.
I have 1 working, and I had 4 working when there was only one box, i.e.
when the handset did not answe
Figured this might be interesting to some on the list. I've written a
small tool to combine the 2 res_monitor created wav files into a single
compressed (IMA ADPCM 8KB/s) stereo file. During playback, you can
select the party to listen to by fiddling with the audio balance.
Email me off-list if
Hello,
From: Shawn <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] sip-msmessenger
Date: 31 Mar 2004 17:03:08 -0500
Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.100:9082
From: ;tag=97442d5b-75b7-4e23-9021-b8605797eb56
To: ;t
I'm having problems with the voicemail prompts. The beginning of each
prompt is garbled. Prompts "edian mail" and "assword" can be
recognized, but when it starts on the short phrases it is completely
garbled. It is almost like the subsequent phrase starts before the
current phrase finishes.
Hi,
Thanks for the reply!
I am still having troubles
I did try:
disallow=all
allow=g726
And still get:
Mar 30 15:49:29 WARNING[-1137677392]: chan_sip.c:2128 process_sdp: No
compatible codecs!
-b
Quoting Michael Manousos <[EMAIL PROTECTED]>:
> Hi,
>
> Bill Hamel wrote:
> > Hi,
> >
> > I a
The password has to match what you have in sip.conf
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roger
Sent: Thursday, 1 April 2004 4:03
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] setting up 7940
Simon Brown wrote:
>Make sure that you
> If you want
> MusicOnHold and Conferencing however, you will need one card for the
timing.
Why - I had used only ztdummy that works for MOH and conf.
uncomment ztdummy in the makefile for zaptel and compile.
/Hans-Henrik Andresen
___
Asterisk-User
Can anyone please help, I can't tell why it will not connect.
I do not know how to read this debug file to were it is wrong.
Thanks
Sip read:
REGISTER sip:192.168.1.101 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:9082
From: ;tag=97442d5b-75b7-4e23-9021-b8605797eb56
To:
Call-ID: [EMAIL PROTECTED]
CSeq
On Wed, 31 Mar 2004, Angus Berry wrote:
> Does anyone hook in to Vonage directly without their Cisco box.. I see
> they have a soft phone out now.
I couldn't get the Vonage softphone working for outgoing calls (incoming
worked ok); need the SIP proxy support mentioned here:
http://bugs.digium.co
I agree... this gem of a link could be more prominent.
I didn't find anything on the WiKi where folks comment about their 3rd
party VoIP providers...did I miss something else?
Does anyone hook in to Vonage directly without their Cisco box.. I see
they have a soft phone out now.
On Wed, 2004-03-3
Are you trying to change the password on the phone or the
password it sends to Asterisk to authenticate with? To change
the password to the phone you need to put:
phone_password: "somepassword"
in either the SIP.cnf or the SIPDefault.cnf, depending how
global you wish to make the password. Remem
I wrote up report on products that caught my interest at the
VON show going on this week in Santa Clara.
http://www.voip-info.org/wiki-VON+Spring+2004+Report
Jim
James H. Thompson[EMAIL PROTECTED]
"Francois Lambert" <[EMAIL PROTECTED]>
At 13:15 31/03/04 +0530, you wrote:
Hi,
How do I detect an Answering Machine in Asterisk.
I saw a post by Francois Lambert on 19 Jan. but am unable to get his email
id.
http://www.mail-archive.com/[EMAIL PROTECTED]/msg02388.html
Can somebody please help?
Tha
On Wed, Mar 31, 2004 at 03:06:21PM -0500, Angus Berry said:
> thanks... this makes for great reading.
>
> Is there a link from Asteriskpbx.org that I missed?
It IS there... Go to support, and it's the first under "user contributed
links". Considering how valuable this Wiki is, it would be nice to
We are looking into using the Linux SMDI code to write a * app module.
have not gotten far.. but we could help... with the Linux SMDI stuff
already written it shouldn't be too hard.
http://rpmfind.net/linux/RPM/contrib/libc6/i386/smdi-0.0.3-1.i386.html
like that
Dave P
>>> [EMAIL PROTECTED]
You would need to turn off call waiting on the phone. I think there's a
menu option or config option for the phone for that.
On Wed, 2004-03-31 at 11:57, Rich Adamson wrote:
> Using the following defnitions with a C7960:
>
> exten => 3001,1,Dial(SIP/3001,15,r)
> exten => 3001,2,Voicemail2(u3001)
"Jason A. Pattie" <[EMAIL PROTECTED]> wrote:
>Is there any possibility to remove the "turnaround" leg or whatever its
>called at the X100P? I'm just thinking of a scenario where none of the
>outgoing signal is ever introduced to the incoming circuit. That way,
>the echo problem simply disappears
thanks... this makes for great reading.
Is there a link from Asteriskpbx.org that I missed?
On Wed, 2004-03-31 at 14:05, Walt Reed wrote:
> On Wed, Mar 31, 2004 at 01:06:03PM -0500, Angus Berry said:
> > It would be useful to have a WiKi on asteriskpbx.com where user could
> > post their experien
Now for the next question. I have an old AT&T Merlin Mail system with a
Brooktrout comcode series 4 cards in it. Could I use them?
Thanks again for your help
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nicolas
Gudino
Sent: Wednesday, March 31, 2004
Hi,
On Wed, 2004-03-31 at 16:00, Hall, Eric M. wrote:
> I have a question for the group.
> To get this running do I need any Digium Cards? I understand I will
> need them to connect to the public phone system. I'm looking at just
> using IP Phones or IP Softphones just to test this app.
You can
Correction to my previous post. You do not need a Digium card for
MusicOnHold just the conferencing...eeuuurgghh
Brian
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Hall, Eric M.
> Sent: 31 March 2004 20:01
> To: [EMAIL PROTECTED]
> Subject: [Asterisk
--On Wednesday, March 31, 2004 2:00 pm -0500 "Hall, Eric M."
<[EMAIL PROTECTED]> wrote:
I have a question for the group.
To get this running do I need any Digium Cards? I understand I will
need them to connect to the public phone system. I'm looking at just
using IP Phones or IP Softphones jus
You can get by without Digium cards if all you want is IPPhones. If you want
MusicOnHold and Conferencing however, you will need one card for the timing.
I am told that you can emulate this hardware feature but with a basic FXO
card costing only 60 quid, why bother.
You can find an excellent descri
On Mar 31, 2004, at 10:12 AM, Steven Critchfield wrote:
On Wed, 2004-03-31 at 11:51, Ryan Thrash wrote:
How do I set configure my voicemail notification so that when I'm left
a voicemail message it:
1) sends an e-mail to my inbox with the voicemail message attached
2) sends a message to my cellphon
>
> A quick hunt around the net shows this Ericsson unit on E-bay as
> H.323 only. The price is good, but I'd rather have SIPactually
> I'd rather have IAX2!
>
> Michael
Yeah.. It is pain to set it up... But it does work very well...
And.. IAX2 box... :) would be very very nice..
Mark? A
Here is how I do the same thing:
exten => 1234,1,Dial(Zap/2,30)
exten => 1234,2,Answer
exten => 1234,3,DigitTimeout,5
exten => 1234,4,ResponseTimeout,3
exten => 1234,5,SetMusicOnHold(random)
exten => 1234,6,BackGround(2)
exten => 1234,7,BackGround(vm-nobodyavail)
exten => 1234,8,Voicemail(21)
Thi
On Wed, Mar 31, 2004 at 01:06:03PM -0500, Angus Berry said:
> It would be useful to have a WiKi on asteriskpbx.com where user could
> post their experiences of 'production' multi-line installs.
>
> I'm 'all ears' to how far folks have scaled their installs especially
> with T1/E1 size systems.
Yo
We're running a T1-based system here (8 incoming lines, 7 of which are in a
rollover group through our telco, 1 of which is our fax line) with a set of
10 Cisco 7940s and a 7960 with the SIP firmware, and so far Asterisk and the
T100P have worked great, modulo a few problems with slightly dodgy har
I have a question for the group.
To get this running do I need any Digium Cards? I understand I will
need them to connect to the public phone system. I'm looking at just
using IP Phones or IP Softphones just to test this app.
Thanks for any help you could give.
_
Hi All.
I'am using Asterisk with SER. I can make call between two internal VoIP
gateways or from na internal to external VoIP gateway. But when I get a
external call, this call hang ups 5 seconds after and I reveive the
following messages
*CLI> -- Executing Dial("SIP/16008-3d17",
"SIP/16007&
We have asterisk server which have hfcs isdn adapter acting as FXO and
mediatrix units as FXS on another site connected with 2mbit link.
The trouble is that when I connect from asterisk server to mediatrix
unit; whatever comes to asterisk is heard OK (no noise, minimum jitter;
absolutely acceptab
I've had my * setup installed with an X100P card for a couple of weeks
and it's great fun! I'm even giving a demo to the local Linux group in
a couple of days.
But I have a snag. I have the X100P on a shared line, and configured to
wait for 20 seconds before answering and doing the
auto-attendan
On Wed, 31 Mar 2004 12:54:02 -0500, Angus Berry wrote:
>A quick search on eBay turned up this 4 port FXO external box for
>US$299:
>
>http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3087347715&category=51279
>
>...anyone know if it's compatible with Asterisk?
>
>Tip: Search eBay for "FXO" and c
So is this the current best price solution for 4 port FXO... and no PCI
slots used?
On Wed, 2004-03-31 at 13:26, Senad Jordanovic wrote:
> Angus Berry wrote:
> > A quick search on eBay turned up this 4 port FXO external box for
> > US$299:
> >
> >
> http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&i
unsubscribe
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Angus Berry wrote:
> A quick search on eBay turned up this 4 port FXO external box for
> US$299:
>
>
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3087347715&category=5
1279
>
> ...anyone know if it's compatible with Asterisk?
Yes.. I can confirm I had it setup and it is working great.
___
In my case, because the server I have decided to use for Asterisk has
most of the slots filled with other cards already. I'm sure my situation
is fairly typical.
Greg
On Wed, 2004-03-31 at 11:22 -0600, John Baker wrote:
> Why not just get a motherboard with 6 pci slots? I don't know your
> set
It would be useful to have a WiKi on asteriskpbx.com where user could
post their experiences of 'production' multi-line installs.
I'm 'all ears' to how far folks have scaled their installs especially
with T1/E1 size systems.
On Wed, 2004-03-31 at 12:09, jc wrote:
> Well, I hope it works because w
On Wed, 2004-03-31 at 11:51, Ryan Thrash wrote:
> How do I set configure my voicemail notification so that when I'm left
> a voicemail message it:
>
> 1) sends an e-mail to my inbox with the voicemail message attached
> 2) sends a message to my cellphone without the message attached
Use procmail
Condition: New Open Box
Warranty: 90 Days
Cost - $130/ea Minimum Order 5pcs
Contact [EMAIL PROTECTED] for details
Cory Andrews
++
b2 technologies
454 Sonwil Drive
Buffalo, NY 14225
++
email - [EMAIL PROTECTED]
voice - 716.630.1555 X22
fax - 716.630.1548
web - www.ValueRe
A quick search on eBay turned up this 4 port FXO external box for
US$299:
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3087347715&category=51279
...anyone know if it's compatible with Asterisk?
Tip: Search eBay for "FXO" and check the box saying search titles and
descriptions.
On Wed, 200
Simon Brown wrote:
Make sure that you have the following (or equivalent in the
SIP.conf
# Line 1 appearance
line1_name: 202
# Line 1 Registration Authentication
line1_authname: "202"
# Line 1 Registration Password
line1_password: "202"
and this in the SIPDefault.conf
# Proxy Registration (0-dis
Using the following defnitions with a C7960:
exten => 3001,1,Dial(SIP/3001,15,r)
exten => 3001,2,Voicemail2(u3001)
exten => 3001,102,Voicemail2(b3001)
exten => 3001,103,Hangup
If someone is on this phone (real conversation) and another call comes in,
the second call goes through the 15 second ti
How do I set configure my voicemail notification so that when I'm left
a voicemail message it:
1) sends an e-mail to my inbox with the voicemail message attached
2) sends a message to my cellphone without the message attached
I get notifications when I've got attachments turned off, but my cell
>How would/do vendor specific phones like cisco 7900's and 3Com VCX 3100
>series phones work SIP to SIP if they download the protocol from the
>Vendor call centers?
At least in the 79xx's case, the phone actually downloads firmware updates
from a TFTP server; by changing the name of the image to u
Tony Wasson wrote:
I have recently noticed that the "Action: Originate" options in asterisk
1.0 CVS has changed sometime between 2/23 and 3/18.
To post a follow up for posterity, 2 tips were suggested when using the
Manager Interface:
1) Make sure to supply Context AND Priority when using
Why not just get a motherboard with 6 pci slots? I don't know your
setup, but you could free up a slot (or two) by buying a motherboard
with a built-in lan (and/or sound).
John
Ariel Batista wrote:
Mark Buckaway wrote:
*This message was transferred with a trial version of CommuniGate(tm)
Pro
Very cool. With the changes to move the MeetMe data into a DB,
would it be possible to add scheduling, password management and
caller limits to the app?
As someone else pointed out this is an almost perfect fit for
our conferencing needs.
Dan
-Original Message-
From: Areski [mailto:[EM
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Brent Franks wrote:
| Agreed. It seems that one out of about every 40 calls has echo problems
| for the duration of the cell. I also notice it a lot more when the call
| is incoming vs. outgoing. But it will still occur on outgoing calls
| occasional
Well, I hope it works because we are installing one in a new office
tomorrow! Ill let you know how it goes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Mangin
Sent: Wednesday, March 31, 2004 5:02 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-User
So in my sip.conf I put many variations of what I thought should go in
there,
finally including (to no avail):
I have had to play this guessing game as well with other codecs/config file
settings in general. And I think this email touches on something that has
been troubling me for sometime.
Scott Laird wrote:
On Mar 31, 2004, at 8:06 AM, Mark Buckaway wrote:
In setting up Asterisk, I'm looking to dump my current phone system
(Nortel Venture). I presently have three POTS lines.
I would use a VOIP provider, but now are presently available in the
Toronto, ON, CANADA area that suppor
Sam,
I too am experiencing a similar problem with the VoicePulse LA
exchange. I have
two VoicePulse DID numbers 212 XXX and 213 XXX. Both numbers terminate
in the same
Asterisk context on my server. DTMF is being passed-through properly on
the 212 number but
not the 213.
I have sent VoicePulse
On Wed, 2004-03-31 at 10:14, Matt Bridges wrote:
> I've configured my [*] to dial the pstn which is working like a charm.
> I've also configured an extension to ring when the PSTN line is ringing
> which is also working brilliantly, but, sometimes it doesn't detect that the
> call has been hungup.
On Wed, 2004-03-31 at 10:06, Yves Chouinard wrote:
> I am projecting to migrate applications written with the Dialogic API
> to Asterisk. There are a few things that I do with Dialogic that I am
> still not sure are possible with Asterisk :
>
> - play a file from an offset (so a user can press a
Yay!! Precisely what I need as well (4-port)...do these rumours include
any word on timeframe?
Greg
On Wed, 2004-03-31 at 08:31 -0800, Scott Laird wrote:
> On Mar 31, 2004, at 8:06 AM, Mark Buckaway wrote:
> > In setting up Asterisk, I'm looking to dump my current phone system
> > (Nortel Ventur
On Wed, 31 Mar 2004 08:31:38 -0800, Scott Laird wrote:
>
>On Mar 31, 2004, at 8:06 AM, Mark Buckaway wrote:
>> In setting up Asterisk, I'm looking to dump my current phone system
>> (Nortel Venture). I presently have three POTS lines.
>>
>> I would use a VOIP provider, but now are presently avail
Matt Bridges wrote:
I've configured my [*] to dial the pstn which is working like a charm.
I've also configured an extension to ring when the PSTN line is ringing
which is also working brilliantly, but, sometimes it doesn't detect that the
call has been hungup.
I've had a look on voip-info and
Mark Buckaway wrote:
> *This message was transferred with a trial version of CommuniGate(tm)
> Pro*
> In setting up Asterisk, I'm looking to dump my current phone system
> (Nortel Venture). I presently have three POTS lines.
>
> I would use a VOIP provider, but now are presently available in the
>
Hello,
Is there any work in progress for supporting SMDI in Asterisk ? if Not, could anyone
tell how to get started implementing it for Asterisk.
Regards,
Tony
$B4dED(B $B?-2p(B wrote:
(B
(B> I got these compile errors while install asterisk.
(B> readline and openssl are compiled using gnu source, and kernel version is 2.4.17.
(B>
(B> Compile errors message is follows.
(B> Someone cleared this problem?
(B> Please, help!
(B>
(B> Regards.
(B
On Mar 31, 2004, at 8:06 AM, Mark Buckaway wrote:
In setting up Asterisk, I'm looking to dump my current phone system
(Nortel Venture). I presently have three POTS lines.
I would use a VOIP provider, but now are presently available in the
Toronto, ON, CANADA area that support user owned hardware
The problems are that you don't have instaled several shell commands like:
(Bpwd
(BThe logs told you in the 7th line.
(B
(BHave a nice day.
(B
(B- Original Message -
(BFrom: "$B4dED(B $B?-2p(B" <[EMAIL PROTECTED]>
(BTo: <[EMAIL PROTECTED]>
(BSent: Wednesday, March 31, 2004
Michael Welter wrote:
You phone isn't getting to the tftp server or the tftp server doesn't
have the required files. Make sure your dhcp server is specifying the
correct IP of the tftp server. You can also check the phone's
parameters to verify the tftp address.
If that is ok then you need a
Hi. I am also doing same thing as you. Currently
my company has a PBX platform which uses Dialogic hardware to control channels.
Now they want their software to work over Asterisk.
I still haven’t started implementing
modifications, as I am still studying Asterisk. I think it would be inte
Net2phone
Kurt
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I've configured my [*] to dial the pstn which is working like a charm.
I've also configured an extension to ring when the PSTN line is ringing
which is also working brilliantly, but, sometimes it doesn't detect that the
call has been hungup.
I've had a look on voip-info and checked the conf file
I set the SISIPRegIntervalo 3600. Wouldn't that mean
to send a registration packet every hour instead of
every miminuter so.
--OR--
Is this the typical reresponseack to the * server when
the ATATAeceive a SIP Notify.
Kurt
__
Hello,
I would like to make sure I am not going to shoot myself in the head, so
I would appreciate if you could tell me if what I am willing to do is
realistic (and will work) or not.
I am willing to replace our office "London" phone system when a new E1
(G703) is installed later this month and I
I am projecting to migrate
applications written with the Dialogic API to Asterisk. There are a few things
that I do with Dialogic that I am still not sure are possible with Asterisk
:
- play a file from an
offset (so a user can press a key to rewind 3 sec., pause,
etc.)
- dynamic volume c
*This message was transferred with a trial version of CommuniGate(tm) Pro*
In setting up Asterisk, I'm looking to dump my current phone system (Nortel
Venture). I presently have three POTS lines.
I would use a VOIP provider, but now are presently available in the Toronto, ON,
CANADA area that su
Greetings
i have been using asterisk with good success with soft phones and I
need to purchase some Sip capable phones to test.
How would/do vendor specific phones like cisco 7900's and 3Com VCX 3100
series phones work SIP to SIP if they download the protocol from the
Vendor call centers?
I w
Reynaldo Simbulan wrote:
Hi Steve,
I am having this problem in which RxFax is still holding the file after
receiving a complete fax. Somehow the zap channel is still active but on the
fax client it was sent successfully.
If you call the line it is still busy.
spandsp-0.0.1k.tar.gz and updated
hi.
(B
(BI got these compile errors while install asterisk.
(Breadline and openssl are compiled using gnu source, and kernel version is 2.4.17.
(B
(BCompile errors message is follows.
(BSomeone cleared this problem?
(BPlease, help!
(B
(BRegards.
(B
(B-
*This message was transferred with a trial version of CommuniGate(tm) Pro*
Does anyone have experience with connect Asterish to Cisco Call Manager 3.x?
What I would like to try to do is to connect my home asterish into the office
Cisco call manager most likely over a VPN from home to the office T
Looks like a cool system.. looking forward to seeing it develop..
Later..
Areski wrote:
Hello Asteriskos,
Screenshot:
http://www.areski.net/asterisk-meetme/about.php
The goals of this application is to control your audience/users in the
conference room. That will allow you to have a visual pres
Terence Parker wrote:
I have posted before but didn't get any replies so i'll ask again in a
more simple way :
Does H323 work on asterisk out of the box? I notice there is already a
channels/chan_h323.c file, but creating an h323.conf file I can't seem
to get H323 working.
Do I have to compi
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