On Thu, 2003-08-14 at 22:41, John Todd wrote:
Since nobody else took the hint, I submitted it as a feature request for SIP.
http://bugs.digium.com/bug_view_page.php?bug_id=104
Personally, this is not high on my I'd love to see this fixed list.
However, many others here are less
Hi,
Checked on the playbacks/voicecalls - only the playbacks have this
problem (I am running Redhat - latest kernel version 2.4.19)
Error Messages (results in stottering audio)
NOTICE[1184091440]: File sched.c, Line 209 (sched_settime): Request
to schedule in the past?!?!
I'm using kernel 2.4.17.
If I try to modprobe zaptel, I receive:
Using /lib/modules/2.4.17/misc/zaptel.o
/lib/modules/2.4.17/misc/zaptel.o: unresolved symbol
remove_proc_entry_Rsmp_5c747b84
/lib/modules/2.4.17/misc/zaptel.o: unresolved symbol
proc_mkdir_Rsmp_b1c61eb3
On Fri, 2003-08-15 at 17:55, [EMAIL PROTECTED] wrote:
I'm using kernel 2.4.17.
If I try to modprobe zaptel, I receive:
Using /lib/modules/2.4.17/misc/zaptel.o
/lib/modules/2.4.17/misc/zaptel.o: unresolved symbol
remove_proc_entry_Rsmp_5c747b84
/lib/modules/2.4.17/misc/zaptel.o:
What should I enable in my kernel to solve this?
The last few certainly point to ppp missing.
On this machine I'm using PPP to connect to my ADSL and to
connect to the pptp server in the company which I work for.
Isamar
___
Asterisk-Users
What does this error message mean?
WARNING[262160]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect
process 2 frames
I've been getting these a lot lately, sound quality seems to have suffered.
I'm using I4L driver with Fritz PCI ISDN card. However, even the regular
echo test sounds a
On Fri, 2003-08-15 at 18:11, [EMAIL PROTECTED] wrote:
On this machine I'm using PPP to connect to my ADSL and to
connect to the pptp server in the company which I work for.
Versions?
Here it's kernel 2.4.20 and ppp-2.4.0-2 (only because of a problem with
pppoatm.so)
.config is:-
CONFIG_PPP=m
Andy Powell wrote:
Can't find the message in a search.. but below is a msg retreved from my
archive..
this is what Mark sent a little while ago
I have no idea if it actually does anything to the card, but on a modprobe I
do get a msg saying it's using CTR21
Andy
I'm in Paris right now
On Fri, 2003-08-15 at 11:42, Richard Scobie wrote:
I guess in his haste to help out the people who were having a problem,
Mark looked at the wrong data sheet when he wrote this patch.
I have corresponded with him and confirmed that this code requires the
Global DAA chipset, which is not
During debugging of H323 trunk side (using Jeremy Macnamara's H323
driver in ~/channels/h323) a couple of things come don't quite work
as advertised...
1/ the following line in extensions.conf explicitly sets the
outgoing caller ID (required in my case for downstream GK
processing..)
exten =
Hi John,
It is not technically possible or it is not yet implemented?
Thanks,
Dan
- Original Message -
From: John Todd [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 15, 2003 1:05 AM
Subject: Re: [Asterisk-Users] How can I know if a user is busy or not
connected?
At
Blind and assisted transfer work with Cisco 7960 phones.
Blind transfer works fine with Budgetones.
As long as you register to Asterisk.
Jamie Carl wrote:
Ok, just been thinking about this and thought I would ask before
trying it out again.
What is the state of SIP transfers? By this I mean
Do you have transfer turn on in zapata.conf?
transfer=yes
Hi,
I cannot use '#' to initiate transfers.
I have tried on different phones (7960, ATA, X-Lite).
When I press '#' during a call, nothing happen.
I have both T and t switches in Dial application.
The transfer function works with Flash key
Email class 101: If you want to start a new thread, START A NEW THREAD.
Do not just change the subject line and remove the old text. Start a new
message. It should be a law, there probably is a RFC for it.
Kernel class 201: If your kernel is compiled with module versions (line
noise looking stuff
Hello list,
my case is as follows:
SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729.
When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the
keypad on the phone.
As suggested by you, I need to configure the SIP1 with out band dtmf mode ,
what is about
Hi,
The problem was not related to ZAP channel. Even between two local SIP phone
was the same.
I have solve it just changing something in the ATA configuration and then
butting back the old value. I still don't know the cause.
Thanks,
Dan
- Original Message -
From: James Sizemore
I am having problems trying to run
asterisk from a telnet session. I am able to su to root and the command
asterisk does not work. Any ideas why this may be occurring? I thought Asterisk
could be configured remotely as well as run remotely?
Thanks in
advance
Steve Lane
H.323 doesn't have an explicit caller*id feature, so any callerid
specific features that have been added are hacks. Since you are using a
gatekeeper why don't you use a type=h323 to specify your H.323 id properly?
[6400047602100]
type=h323
secret=securepassword ; optional
Find me on IRC
Please explain, what do you mean by does not work. It looks like a path
problem. Instead of using su use su -.
su does not initialize the environment for the user you are suing as,
but su - will. /sbin and /usr/sbin a special path's that are only in
root's environment.
Hope this help, I
On 14/8/03 12:53, WipeOut . [EMAIL PROTECTED] wrote:
The highest quality codec is ulaw or alaw (otherwise know as G.711).. These
are the same as what comes in on your PSTN line..
If you want high Quality voice prompts your best bet os to record them on a PC
with a good quality mic and then
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Please explain, what do you mean by does not work. It looks like a
path
problem. Instead of using su use su -.
su does not initialize the environment for the user you are suing
as,
but su - will. /sbin and /usr/sbin a special path's that
I was interested in getting the highest quality over a normal phone line
because I want users to be able to record their messages at the highest
quality. They do it not me hence my question about the highest quality
codec.
If for example they used VoIP software on a computer could they get
Forgive me as my Linux and UNIX skills have not been utilized in over 6
months due to my job situation. I thank you for the suggestion of the
- after the su command. It worked and I am slowly regaining my memory.
Again I am sorry to ask such elementary questions, but the simple info
helps me quite
On 15/8/03 18:57, WipeOut . [EMAIL PROTECTED] wrote:
I was interested in getting the highest quality over a normal phone line
because I want users to be able to record their messages at the highest
quality. They do it not me hence my question about the highest quality
codec.
If for example
Personally I'd use ssh rather than telnet
Andy
*** REPLY SEPARATOR ***
On 15/08/2003 at 12:21 Steve Lane wrote:
I am having problems trying to run asterisk from a telnet session. I am
able to su to root and the command asterisk does not work. Any ideas why
this may be
Thanks for the fast reply.
I assume you mean that alaw or ulaw is what the carrier delivers. Because my
customers use the phones they have I do not have control over the carrier
they use, I just meant it to mean all carriers. As in what is the standard
format they deliver in.
I assume
Repeat after me. Telnet BAD ssh good!
bkw
On Fri, 15 Aug 2003, Andy Powell wrote:
Personally I'd use ssh rather than telnet
Andy
*** REPLY SEPARATOR ***
On 15/08/2003 at 12:21 Steve Lane wrote:
I am having problems trying to run asterisk from a telnet session. I
On 15/8/03 20:15, WipeOut . [EMAIL PROTECTED] wrote:
Thanks for the fast reply.
I assume you mean that alaw or ulaw is what the carrier delivers. Because my
customers use the phones they have I do not have control over the carrier
they use, I just meant it to mean all carriers. As in what
Yes you probably right.
Test and see.
So how do I get and use each codec to test them. I understand some are under
tight copyright control. Do they have testing variations rather than buying
first, find it it does not work, and your stuck with the cost?
Thanks for you help.
As
I was thinking of a hotline set up something like this:
FXO --*--IAX--*--FXS
The dialtone has to be provided by the remote end and flash hook has to be
transparent
Anyone have experience with hotlines on *? Would this work?
cheers,
darren
---
Checked by AVG anti-virus system
Hi
I am using asterisk with a Quicknet lineJack card. I am trying to get a
proof of concept demo together before real deployment. I have a couple of
qustions.
1) Can I use the codecs that are on Quicknet card (G.723, etc, etc). I
tested it and I cannot use any codec in hardware, the only
I guess my big question is: is it possible to have extensions mapped to
people, not to phones?
Yes, you just need to link the user/extension to a technology/channel
when logged in, and to a bogus value when not so that your dial will
fail quickly and fall through to voicemail. Also you
http://www.digium.com/index.php?menu=asterisk_g729
bwk
On Fri, 15 Aug 2003 [EMAIL PROTECTED] wrote:
Hi
I am using asterisk with a Quicknet lineJack card. I am trying to get a
proof of concept demo together before real deployment. I have a couple of
qustions.
1) Can I use the codecs that
Can someone tell me where I can get information on how to
configure the sip.conf file to register a soft phone?
I tried the following entry and the phone will not register:
[xten1]
type=friend
username=xxx
secret=ppp
host=dynamic
I am using the X-Ten Lite soft phone,
but I
Detecting what type of error or call result was produced by the
Dial application has not yet been implemented, though it is
desperately (IMHO) needed to allow the dialplan to more reasonably
direct calls to the correct subsequent context. If I could code it,
I would, but I can't, so I
I am glad to hear this DTMF issue has been resolved. With this in mind I
would like to know if there is an index of Asterisk Consultants who
build/sell/support Asterisk IP PBXs in North America.
Xten on occasion receives sales inquiries for IP PBXs and I would like to
refer this business to the
On Fri, 2003-08-15 at 17:40, Devon Henderson wrote:
I guess my big question is: is it possible to have extensions mapped to
people, not to phones?
Yes, you just need to link the user/extension to a technology/channel
when logged in, and to a bogus value when not so that your dial will
On Fri, 2003-08-15 at 12:42, Adams, Gavin wrote:
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Please explain, what do you mean by does not work. It looks like a
path
problem. Instead of using su use su -.
su does not initialize the environment for the user you are suing
as,
Quicknet does not offer G.729. Make sure that you use the nixj driver
for the quicknet card. It's better than the old driver shipped with the
kernel. As soon as the new driver is done, we plan to send it to the
kernel folks so it can replace the old one.
I believe that You can get a 729
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