If you see nothing with full verbosity and SIP debug turned on, the Asterisk SIP channel gets nothing.
The reason why we always mix in NAT with questions like yours is that in 90% of the
cases, NAT
is the problem. It's just a standard response, like when Microsoft support tells you
to reinstall
Hello,
I have one question again. I checked archive and I found that somebody
before me asked this question already.
But no responses for this posting.
http://lists.digium.com/pipermail/asterisk-users/2003-September/020065.html
So, is it supported or no? If yes, what I need to configure?
Thank
Dear all,
My GS ATA-286, which otherwise work well, seem to be unable to ring a
fax (or at least, some kind of fax). The fax basically doesn't detect
the ring.
I measured with a volt meter about 45V during the ring pulse out of
the ATA. This looks fairly low to me (supposed to be in the 70V+
Hi
I have an ISA Diva 2.0 ISDN card and i am using i4l
as well, and i use the same calling method, it workd for me.
Can u show your modem.conf?
remember to use in modem.conf
driver=i4l
and
group=1
msn=0
incomingmsn=XXX ; your
incoming numbers
device = /dev/ttyI0
device = /dev/ttyI1
it is solved, thanks!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
dfmSent: Monday, February 23, 2004 9:14 AMTo:
[EMAIL PROTECTED]Subject: Re: [Asterisk-Users] 2
questions about ISDN BRI
Hi
I have an ISA Diva 2.0 ISDN card and i am using i4l
as well, and i use the
Thank you to Olle Johansson, Philipp von Klitzing, and others who
suggested approaches to the problem.
To summarise what we did and how we ended up solving the problem:
Situation:
1. Grandstream phone behind NAT box.
2. Asterisk not behind NAT (with static IP).
3. Phone cannot register.
Well after a bit more googling, I've found the quick nasty fix to this
problem. Users on the Norstar extensions need to dial Feature 808 to enable
Long Tones so that when they press a key on their keypad, it's passed
correctly to the Analog Terminal Adapter.
I call this a partial solution, since
John,
You are now advertising your EMEA company in your signature block. Maybe
I missed an email that explains the EMEA pricing and availability. Could
you please give an update via the list as to the status of your product
availablity, pricing and delivery times in Europe? The ordering
Soren Rathje said:
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 22, 2004 8:52 PM
Subject: Re: [Asterisk-Users] SIP extension busy when not available ??
Although the current logic does not require a sip phone to
The only adapter that I know of that allows you to modify the ring voltage
is the Sipura analog SIP adapter. I was able to get my old fax machine to
answer after jacking up the ring voltage to 90V. http://www.sipura.com
MATT---
-Original Message-
From: Nicolas Bougues [mailto:[EMAIL
Title: Re: [Asterisk-Users] New Wiki page: Dimensioning an Asterisk system
Is the article correct in saying:
g729 codecs licenses
can be purchased for Asterisk (not for SCSI systems!)
I thought people had this working on SCSI
now?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
Jose Quinteiro wrote:
I live at sea level, and have never seen a woodpecker going at any telco
equipment, but have a 60Hz hum on my POTS line through my Adtran 750.
It goes away if I pick up the telephone I have cross-connected on the
same line. Could it be the same problem (i.e., tip-ring
I think international number dialed through voicepulse have to start
with 011... (even if you're located in another countery). I asked them
about that once, and that's what works for me (We've been dialing Spain
and Germany recently, but never Japan)
HTH,
Matt
--__--__--
Message: 4
In the UK it's 00 then the country code. So a call from the UK to my
phone would be 0013036742575.
Miie
Matt Lawson wrote:
I think international number dialed through voicepulse have to start
with 011... (even if you're located in another countery). I asked them
about that once, and that's
On Sunday, February 22, 2004 2:04 AM, James H. Thompson
[SMTP:[EMAIL PROTECTED] wrote:
Other vendors are seeing the benefits of open source:
From: http://www.pingtel.com/a_opensource.jsp
Announcing the emergence of an enterprise-class open source IP
PBX
Please cancel my previous post.
Matt Lawson wrote:
I think international number dialed through voicepulse have to start
with 011... (even if you're located in another countery). I asked them
about that once, and that's what works for me (We've been dialing Spain
and Germany recently, but
That's what I'm trying to get at. *normally* you expect to dial 00 but
when you're using voicepulse, Asterisk needs to start all international
number with 011. Think of it this way, in VoicePulse's mind, you're
always dialing from the US. Of course the user will try dialing 00
because
Don Pobanz wrote:
On Sunday, February 22, 2004 2:04 AM, James H. Thompson
[SMTP:[EMAIL PROTECTED] wrote:
Other vendors are seeing the benefits of open source:
From: http://www.pingtel.com/a_opensource.jsp
Announcing the emergence of an enterprise-class open source IP
PBX
Don == Don Pobanz [EMAIL PROTECTED] writes:
Don I do not know what 'Linux-style subscription license' means.
That one stalled me for a bit, too. Based on their ad copy they
are offering annual support contracts for the system, but releasing
the code itself under some free/open license. (I
I always keep a terminal window open with top running for my asterisk
servers. Since we've had Asterisk in production, for about 9 months, I've
noticed with every platform and every card we've tried that the load average
will be going along at about 0.1 to 0.5 with about 30 channels(15 SIP -
Zap
Hello.
I've just recently purchased the Asterisk Developers Kit so we can
figure out how to get away from our Nortel system and go to IP based
phones. I have a RH 9 box loaded with Asterisk (a very recent cvs download).
Either way, I can call the asterisk box and get their demo playing fine.
Snom TAPI integration is a joke...
Andy
*** REPLY SEPARATOR ***
On 22/02/2004 at 21:47 Peer Oliver schmidt wrote:
Hi,
anyone here running SNOM phones with TAPI integration with Outlook?
Any other hardware phone with some TAPI integration?
rgds
pos
Jason,
Include your sip and extensions files so people can take a look.
T
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Sent: Monday, February 23, 2004 10:25 AM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] An example
WipeOut wrote:
This is an interesting statement in the press release..
SIPxchange, the industrys first open source based enterprise
communications suite, is grounded in the concept that a community of
ideas provides a more fertile ground for innovation, progress and
product development.
I
I'm writing an application for asterisk (really just a set of access
commands to the builtin API), and I notice that a lot of existing
applications are not thread-safe. Should they be? Should mine be?
Thanks,
--Ernest
___
Asterisk-Users mailing list
Andy Powell wrote:
Snom TAPI integration is a joke...
Would you mind elaborating a bit on this? Is the future implemented, but
does not work, or is it not implemented at all? Or something else?
Thanks
rgds
pos
___
Asterisk-Users mailing list
[EMAIL
On Monday 23 February 2004 10:15, Ernest W. Lessenger wrote:
I'm writing an application for asterisk (really just a set of
access commands to the builtin API), and I notice that a lot of
existing applications are not thread-safe. Should they be? Should
mine be?
Could you elaborate, please?
At 08:31 AM 2/23/2004, you wrote:
On Monday 23 February 2004 10:15, Ernest W. Lessenger wrote:
I'm writing an application for asterisk (really just a set of
access commands to the builtin API), and I notice that a lot of
existing applications are not thread-safe. Should they be? Should
mine
Hello,
I was curious if there was any way to play a tone on Attended transfer
once it bridges the party being transferred to the destination?
Basically what is happening now is:
1.) A caller calls in using a zap channel
2.) Call is sent to SIP Polycom Phone - Receptionist
3.) Receptionist
On Mon, 2004-02-23 at 10:55, Ernest W. Lessenger wrote:
At 08:31 AM 2/23/2004, you wrote:
On Monday 23 February 2004 10:15, Ernest W. Lessenger wrote:
I'm writing an application for asterisk (really just a set of
access commands to the builtin API), and I notice that a lot of
existing
At 09:14 AM 2/23/2004, you wrote:
Why would you program something that isn't thread safe? From what I can
tell, it isn't much extra effort to do things the right way instead of
debuging crap later.
I wouldn't, and generally don't. But sometimes (rarely) you need to include
functions that aren't
On Mon, 2004-02-23 at 09:19, mattf wrote:
I always keep a terminal window open with top running for my asterisk
servers. Since we've had Asterisk in production, for about 9 months, I've
noticed with every platform and every card we've tried that the load average
will be going along at about
On Mon, 2004-02-23 at 11:22, Ernest W. Lessenger wrote:
At 09:14 AM 2/23/2004, you wrote:
Why would you program something that isn't thread safe? From what I can
tell, it isn't much extra effort to do things the right way instead of
debuging crap later.
I wouldn't, and generally don't. But
Thanks for the response. I plan on trying Slackware on my backup/test
asterisk server when I have a new backup server ready in a few weeks. I've
noticed in some database machine testing that Slackware starts up in about
half the time of RedHat and doesn't have all of that Redhat junk either.
I'll
Looking through the Wiki and mailing list, I didn't see an answer to
this.
Is there a way to set the minimum voice mail message size? Hangups seem
to generate 4 to 5 second messages. If I set a min to 6 or 7 that should
eliminate most of these.
The main voicemail app also seems kind of thin.
Timothy,
I have minimally modified the demo files that came with Asterisk, so
what is posted below is most of the comments and the demo section
removed from the config files.
Thanks!
; SIP Configuration for Asterisk
;
[general]
port = 5060; Port to bind to
bindaddr = 0.0.0.0
When compiling Asterisk for a dual XEON based
system are there any caveats or "switches" that we need to be aware
of?
Hello,
I am interested in running small busines in telecommunication with minimum
expenses and investment. Can Windows operating be used for this purpose.
Thank you all.
Regards,
Yaseen
From: Michael Manousos [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject:
On Mon, 2004-02-23 at 18:42, mattf wrote:
Thanks for the response. I plan on trying Slackware on my backup/test
asterisk server when I have a new backup server ready in a few weeks. I've
noticed in some database machine testing that Slackware starts up in about
half the time of RedHat and
When placing a call from Sipura SPA 2000 to other extensions, for some
reason
dialled extension keeps ringing even though SPA 2000 hangs up the call.
Asterisk does not end that call until it is not answered by dialled
extension.
Anyone has experienced similar problem?
Dear All
i have modified the app_queue.c with the patch app_queue_patch_1_07
from bug
track to use with Asterisk 0.7.2 i have try it and seems to work :-)
I hope to help someone
Bye
Dimi
app_queue.c_queue_patch072.tar.gz
Description: application/tgz
On Monday 23 February 2004 12:56 pm, Khalid Yaseen wrote:
Hello,
I am interested in running small busines in telecommunication with
minimum expenses and investment. Can Windows operating be used for
this purpose. Thank you all.
Regards,
Yaseen
Haha, that's funny!
Unless of course you
Try moving your sip phone into its own context, instead of default (I use
sip) and create a [sip] section in your extensions.conf
Add a sepcific extension to test your outgoing, like :
exten = _5,1,Dial,Zap/1/800551212
T
-Original Message-
From: [EMAIL PROTECTED]
I have been trying to set up three * servers to use IAX between them and
am a bit lost as to the finer detail of the config files. I have read
the wiki and it has not made things better.
Here is my problem;
I create a section like this on each machines:
[othermachine-1]
type=friend
Title: Pickup
The extension for Pickup seems to be *8#, but I cannot find it anywhere in any configuration file. Is this a hard wired extension? Are there other hard wired extensions? If so, is there a list? What priority do they have? Is there any way to change them or map additional
Hi,
I wish my IAX connection negotiates codecs in the following order:
1) speex
2) gsm
3) alaw
Is it possible? I tried and I detected * selects gsm prior to speex no
matter the order I write my iax.conf allow command.
Daniel
___
Asterisk-Users
It works for me for internal calls, and for setting up calls over a PRI.
However, there are problems with overlap dialling when using an X100P
analogue card as * does not seem to buffer digits correctly. I would
recommend not using overlap sending with SIP phones til those issues ar
fixed.
You cannot specify the order of codec selection with Asterisk
On Mon, 2004-02-23 at 13:03, Daniel Bichara wrote:
Hi,
I wish my IAX connection negotiates codecs in the following order:
1) speex
2) gsm
3) alaw
Is it possible? I tried and I detected * selects gsm prior to speex no
I have 2 lines setup. One is the house line, the other a business line.
What I'd LIKE to do, is if a house extension dials out, it selects the
house line to dial out on, but if the house line is busy use the
business line.
Ditto with the office extension, but reverse.
Using distinctive ring on
Really?
Did you try
disallow=all
Allow=speex
Allow=gsm
Allow=alaw
?
T
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Monday, February 23, 2004 2:21 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Codec Order / Preference
Hello all,
I finally got around to installing my Dev Kit Lite. I did the install
yesterday from the latest CVS. I am receiving an error that does not let *
start up. When I go through the procedure to load the modules, I get the
following error after running ztcfg.
Channel map:
Channel 01: FXO
Ed Devine wrote:
When compiling Asterisk for a dual
XEON based system are there any caveats or "switches" that we need to
be aware of?
Well, for zaptel hardware you need to uncomment the SMP entry in the
zaptel Makefile. Also I would turn off Hyperthreading (in the bios). It
may
I've looked through a lot of different pieces of documentation regarding
*'s ACD functionality. Is there any one place in particular with a good
amount of documentation on it?
Thanks
Mark
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Mon, 23 Feb 2004, reseaux wrote:
Dear All
i have modified the app_queue.c with the patch app_queue_patch_1_07
from bug
track to use with Asterisk 0.7.2 i have try it and seems to work :-)
I hope to help someone
Bye
Dimi
What Bug number is this?
--
Vice President of
Can anyone
help me, (after a two day search, also on the mailing list)
I have the
following situation:
Asterisk
works fine, until I added a FXO card. (Digium)
When I
tried to call to the pstn I have the following error
Executing
Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack
Regovich, Timothy wrote:
Really?
Did you try
disallow=all
Allow=speex
Allow=gsm
Allow=alaw
Yes and it did no work.
?
T
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Monday, February 23, 2004 2:21 PM
To: [EMAIL PROTECTED]
On Mon, 2004-02-23 at 17:10, Wim Venneman wrote:
Can anyone help me, (after a two day search, also on the mailing list)
I have the following situation:
Asterisk works fine, until I added a FXO card. (Digium)
When I tried to call to the pstn I have the following error
Executing
Make sure you run a ztcfg after you do a modprobe.
ztcfg will configure (or bring up) the zap channels on zaptel interface
cards. Do this before starting * and after the modprobe.
(You may also do a ztcfg -v to see whats configured)
- Brent
-Original Message-
From: [EMAIL PROTECTED]
That still does not tell Asterisk the ORDER of the codec selection.
On Mon, 2004-02-23 at 13:28, Regovich, Timothy wrote:
Really?
Did you try
disallow=all
Allow=speex
Allow=gsm
Allow=alaw
?
T
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I use ChanIsAvail() to check to see if the phone is connected at the top
of the dialplan for that extension. This works for IAX2 and SIP channels
but not for MGCP.
If you are interested in the actual code I can send it to you from home
tonight.
Robert
Thank you, yes please...
Well,
Hi!
I'm trying to record som voiveprompts, and I've created a directory se in
/var/lib/asterisk/sounds - in that directory I've put files like
vm-intro.gsm, vm-the-person.gsm and do on. And if I use SetLanguage(se) I
hear my own voice prompts!
But wehre should I place the digits I've recorded? -
Assuming that getting H323 to work over NAT is almost really hard
What is
about having both SIP clients venid different NATs ¿ is it posible or as
hard as H.323?
Thanks!
Marc.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Dear Geert
I use * with 1 TE400P on Dual Xeon with 1GByte of RAM HT everyday with
little
30 channels load of calls at time, can you give me more info about problem in
this kind of configuration?
thanks
Dimitri
On Monday 23 February 2004 19:31, Geert Nijpels wrote:
Ed Devine wrote:
Lars Fredriksson wrote:
Hi!
I'm trying to record som voiveprompts, and I've created a directory se in
/var/lib/asterisk/sounds - in that directory I've put files like
vm-intro.gsm, vm-the-person.gsm and do on. And if I use SetLanguage(se) I
hear my own voice prompts!
But wehre should I place the
On Monday, Fedbruary 23rd Olle wrote:
Change no to se (who cares about norwegian :-) ) and you'll be ok.
And remember to report this to bugs.digium.com - tack!
Hey, now!!
Thorsten
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I have a x101p and I can't seem to get ztmonitor to work on it. I've
tried it on 2 different machines. One with a SBLive! card and the other
with a AMD-768 [Opus] Audio (rev 03) chip. Neither machine give me a
graph in ztmonitor 1 -v mode. If I run ztmonitor without the -v I
get:
reseaux wrote:
Dear Geert
I use * with 1 TE400P on Dual Xeon with 1GByte of RAM HT everyday with
little
30 channels load of calls at time, can you give me more info about problem in
this kind of configuration?
thanks
Dimitri
I never did experience problems that could be directly
Made changes:
1)
musiconhold= default
channel = 1
2)
reboot
modprobe wcfxo = ok
ztcfg -v
result = 1 channel configured
Try to dial, still the same problem. (error)
Wim
- Original Message -
From: Brent Franks [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 23, 2004
|I cannot call out with my SIP phone though. It'll dial, ring my cell
|phone twice and then give up and complain that its busy. Even if I try
|to answer the cell phone during the first ring.
|
|Does anyone have a config they could share with me on how to make this
|setup work? This sounds like it
hi everybody,
just went into some trouble (again!!) while I was
trying to make a call between two (isdn)phones connected to my hfc-s card. I am
running junghanns.net's hfc-bri-driver. the call is terminated after a few
seconds.
anyone else got this to work? btw: I am using a
NTBA as
Ah! I just checked out the latest ztmonitor out of cvs and it works
just fine.
...Jeff
On Mon, 2004-02-23 at 12:51, Jeff Gustafson wrote:
I have a x101p and I can't seem to get ztmonitor to work on it. I've
tried it on 2 different machines. One
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E.
Johansson
Sent: Monday, February 23, 2004 9:41 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VM: Multilanguage and digits
Lars Fredriksson wrote:
Hi!
I'm trying to record som
Wim, I made some changes to your Zapata.conf and zaptel.conf config
files below.
Hope this helps.
Also, do a less /proc/interrupts and see if the card is on it's own IRQ.
- Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wim Venneman
Sent:
From Posts on this list on Sat. w/ the subject Voicemail brought to
light that there is a patch for some more advanced VM features after a
message is left.
http://bugs.digium.com/bug_view_page.php?bug_id=156
On Mon, 2004-02-23 at 12:56, Walt Reed wrote:
Looking through the Wiki and mailing
I am trying to get a Cisco 12SP phone to work with *.
I do not have call manager.
When start * and turn skinny debugging on I get this on the console:
--
-- Starting Skinny session from 192.168.1.202
Recieved AlarmMessage
I have the following situation:
my cisco 7940 attached to asterisk
i use asterisk as voice mail, conference room, music on hold and so on.
also i have cisco 5350 and use it as PSTN gateway.
-
As i know asterisk able to forward g729 frame.
I enable in asterisk sip,conf
Thanks for the help !
Made changes, still the same message.
I have two NIC's with IRQ 11
The FXO card has IRQ10 (and no other card has IRQ10)
Wim
- Original Message -
From: Brent Franks [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 23, 2004 10:21 PM
Subject: RE:
Soren Rathje said:
I use ChanIsAvail() to check to see if the phone is connected at the
top
of the dialplan for that extension. This works for IAX2 and SIP channels
but not for MGCP.
If you are interested in the actual code I can send it to you from home
tonight.
Robert
Thank you,
The route of my call is:
gs101--asterisk--iaxtel--asterisk--gs101
I have 2 g729 from Digium and calls to iaxtel can only be in gsm format.
The GS101 phones are set to use g729, then 711ulaw.
However when the called GS phone is picked up the connection is
terminated. These are the console
I remember we had something one or two years ago, but I remember that was
not what I was dreaming of.
Sorry we are not so good in implementing Windows-stuff... Maybe has someone
out there a template for TAPI? Something for someone who never did something
with COM or DCOM or .net or whatever...
Just done a cvs checkout on 2 different machines 1 running Mandrake 9.2
with a 2.4.25 kernel the other Mandrake 10.0 and 2.6.3
the most 10.0 fails at this point
chan_zap.c: In function `handle_init_r2_event':
chan_zap.c:4773: error: too few arguments to function `zt_new'
make[1]: *** [chan_zap.o]
On Mon, 23 Feb 2004, Christian Stredicke wrote:
I remember we had something one or two years ago, but I remember that was
not what I was dreaming of.
Sorry we are not so good in implementing Windows-stuff... Maybe has someone
out there a template for TAPI? Something for someone who never
Senad Jordanovic wrote:
When placing a call from Sipura SPA 2000 to other extensions, for
some reason dialled extension keeps ringing even though SPA 2000
hangs up the call.
Asterisk does not end that call until it is not answered by dialled
extension.
Anyone has experienced similar
On Mon, 23 Feb 2004, Dave Cotton wrote:
Just done a cvs checkout on 2 different machines 1 running Mandrake 9.2
with a 2.4.25 kernel the other Mandrake 10.0 and 2.6.3
the most 10.0 fails at this point
chan_zap.c: In function `handle_init_r2_event':
chan_zap.c:4773: error: too few
Jim Sneeringer wrote:
The extension for Pickup seems to be *8#, but I cannot find it
anywhere in any configuration file. Is this a hard wired
extension? Are there other hard wired extensions? If so, is there a
list? What priority do they have? Is there any way to change them
or map
Hi,
Could anyone tell me if asterisk supports multicast? And if so, what
type? And if not, are there any plans to implement one in the forseeable
future?
Thanks,
Jason
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I tend to agree with Christian, imho if there is something a joke
then it is TAPI. There are lot of service creation techniques,
be distributed REFER-based or centralized B2BUA-based which
take no additional .*APIs.
-jiri
On Mon, 23 Feb 2004, Christian Stredicke wrote:
I remember we had
hi christian,
have a look at http://www.julmar.com/. TSP++ version 2 is a opensource,
GPLed library for creating a tapi service provider.
I think this is a good point to start. I was just dreaming of having such a
baby for use with asterisk* via it's manager function.
bye
thorsten
-
You can't do what you're trying to do. Asterisk isn't forwarding the g729
when you check voicemail. The voicemail is part of Asterisk so, it has to
be able to speak the codec that you're using.
John
Alex Ovcharenko wrote:
I have the following situation:
my cisco 7940 attached to asterisk
hi everybody,
does the zaphfc driver support the *8#, *78#, *72#,
... functions when running in NT-mode?
thanks...
bye
thorsten
Is it possible to have nested include statements in iax.conf? Example:
[access-1]
; Can only use resouces in this context.
[access-2]
;can use resources in this context and access-1
include = level-1
[access-3]
;can use resources in all three contexts.
include = level-2
If we can't do this,
I have Suse 9.0 with gcc3.3.1 (didn't have any
problem with the previous version of gcc )and when I run make install I get the
following error:
/usr/lib/gcc-lib/i586-suse-linux-/3.3.1/../../../.../i586-suse-linux/bin/ld:
cannot find -lz
Any help would be appreciated.
Dan
I'm tring to register my ATA to * and I getting the following message:
Feb 23 18:13:04 NOTICE[1125329600]: chan_sip.c:5405 handle_request:
Registration from 'sip:[EMAIL PROTECTED] user=phone' failed for
'xxx.xxx.xxx.xxx'
I don't know what's wrong an why it register as user=phone???
Coul some
Make sure that zlib is installed and its location is in your
LD_LIBRARY_CONFIG path (or /etc/ld.so.conf, at least on RH it's that file, I
assume that SuSE is the same). This package would be on your SuSE CD(s),
it's pretty much a base Linux package.
Greg
- Original Message -
From: Dan
I finally figured it out. Had to install zlib-devel
package.
sorry for the posting, but it was driving me
nuts.
- Original Message -
From:
Dan Fernandez
To: [EMAIL PROTECTED]
; [EMAIL PROTECTED]
Sent: Monday, February 23, 2004 8:07
PM
Subject:
Dave Cotton wrote:
Just done a cvs checkout on 2 different machines 1 running Mandrake 9.2
with a 2.4.25 kernel the other Mandrake 10.0 and 2.6.3
the most 10.0 fails at this point
chan_zap.c: In function `handle_init_r2_event':
chan_zap.c:4773: error: too few arguments to function `zt_new'
Costa Tsaousis wrote:
Also I would turn off Hyperthreading (in the bios). It
may cause problems.
What problems? Are these digium H/W specific, asterisk specific or
generaly Linux problems?
I don't know if The HT problems are generic, or something quirky in the
Zaptel drivers. However, if
Andres,
thanks for your reply. I beg to disagree, here are the arguments:
1) Having INFO is imho a useful thing: it allows elements out of the
media path to control DTMF-based service logic. Otherwise, you
will end up processing media which affects bandwidth and latency
noticably and
Hi!
The extension for Pickup seems to be *8#, but I cannot find it anywhere
in any configuration file. Is this a hard wired extension?
Yes, but you can override it in extensions.conf.
Are there other hard wired extensions? If so, is there a list? What
priority do they have? Is there any
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 23, 2004 10:45 PM
Subject: Re: [Asterisk-Users] SIP extension busy when not available ??
I use a macro to define the extensions. In this way I only have to enter 1
line per actual extension.
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