Henry Devito:
exten = 3,8,Dial(sip/${destination}D{$pin})
^^
Awoogah. Awoogah.
Nick.
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To
Title: Frequency Shift
Hello,
I am using * as a SIP proxy with several SIP clients. The SIP clients are SJPhone Soft phones. All clients are inside a firewall and the Server is inside too. All is working fine, but the speech sounds like Micky Mouse. If you feed one client´s (Mic) input with a
I am reading the manual from Bugetone 101 and found on page 19, the setting
for [8] SIP SP-1 till SP-9
That would be nice! Could leave the FWD number in place, while I test my
Asterisk setup !!
However, I did not find out how I can setup SP-2 ~ SP-9
(Only configured SIP server(s) are
Hi there,
I'm trying to set up a small asterisk box for our company, and am
using a TDM400P with an FXO module in it for one of the external
PSTN lines.
I'm having problems getting Asterisk to detect the remote caller
hangup; when a call is received, I get the following messages
on the Asterisk
hi
when looking into the sipfriends table (using mysql sipfriends from
asterisk cvs version -r v1-0), I see timestamp and ipaddr set to
0/NULL. When looking into the CDR, the user has dialled out recently.
Also sip show peer xxx shows no data.
How can this be true?
roy
I must admit I live in perpetual fear of forgetting to switch of html or rtf
formatting (useful for work) and top posting. I can understand the issue with
the former but can see absolutely no reason why top posting is such a problem.
In fact I'd far prefer it. I get to my e-mail in batches
On Wed, 2004-11-10 at 22:25 -0600, Rich Adamson wrote:
Just a quick FYI for the Aastra/Sayson 480i SIP phone
Just received one and now have it running with *.
- Unit came with SIP v1.0.0.34 Release code 0035-00-00 installed. No
CDROM shipped with the unit, and a quick look at
Hi,
(B
(BDate: Thu, 11 Nov 2004 08:42:16 +0200 (SAST) [zone:-], [EMAIL PROTECTED]
(Bmentioned in msg: Re: [Asterisk-Users] quasi-skype channel for Asterisk?
(Bthat ...
(B
(B On Wed, 10 Nov 2004, Kuniyoshi Murata wrote:
(B
(B http://www.pcphoneline.com/skype
(B
(B If I have a spare
George Gardiner [EMAIL PROTECTED] writes:
So that I can understand the almost religious fervour on this point
could someone please explain to me why top posting is so hated!!
Because there's such an enormous amount of communication one would
like to take part in, and not enough time. The
From: Tom Ivar Helbekkmo [mailto:[EMAIL PROTECTED]
Sent: 11 November 2004 09:38
My default is to move on; only if your posting quickly establishes that
it is, in fact, interesting to me, will I read it. To put it bluntly:
if you can't be bothered to make an effort to communicate, what you say
Hi Henry,
I have found your message in the mailing list archive from October where you
describes compiling problem with the app_conference.
Now I have exactly the same problem with it.
Have you found any solution of this problem?
Link to our message:
Oh, that's a great idea, Tom. Let's have everyone operate to your exacting
standards. I can appreciate that not everyone did their degree in mail list
etiquette and have lives to live and so want to be economical with their
time.
So for my part I scan emails top, bottom or otherwise posted and
hi,
can someone give me any hints if the old german ISDN protocol '1TR6' is
supported by asterisk.
we have a potential customer who has an existing conventional PBX which
has to be extended by an asterisk server. unfortunately this existing
PBX speaks 1TR6 on it's ISDN ports.
regards
frank
Hi,
I havent received many replies so i was just wondering again if
anyone has any thoughts of the 404 call not found issue.I have only a
very basic configuration which can be seen below in the original
email. I have since modified this so that each client (i.e. 2000 and
2001) have
Hello,
try this document (from the wiki):
http://www.astmasters.net/stuff/X-Lite-and-Asterisk.pdf
setting the auth param and the canreinvite and reinvite might help.
-yair
On Thu, 11 Nov 2004 10:38:55 -, Ashling O'Driscoll
[EMAIL PROTECTED] wrote:
Hi,
I havent received many replies
is grandstream still in business??
- Original Message -
From: Ronald Wiplinger [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Thursday, November 11, 2004 3:49 AM
Subject: [Asterisk-Users] Grandstream BugeTone 101 - Multi-Server
You can change the dial plan in the .cfg file if you have that on a tftp
server.
Julian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez
Sent: 11 November 2004 08:57
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On Thu, 2004-11-11 at 11:38 +0100, Frank Sautter wrote:
can someone give me any hints if the old german ISDN protocol '1TR6' is
supported by asterisk.
we have a potential customer who has an existing conventional PBX which
has to be extended by an asterisk server. unfortunately this existing
You will always want a good over-capacity power supply for an AMD server(or
any production server for that matter) I always buy nice heavy 500W+ power
supplies for all of my servers whether they be AMD or Intel-based. For AMD
I've used TTGI power supplies mostly and for Intel I usually use Antec.
well i have an icon diva quadbri card and i already tried uploaded the
1TR6 firmware, which seems to work so far.
the problem is, that the capi module and therefore chan_capi do not load
correctly.
Patrick wrote:
I know my Eicon Diva Server BRI card supports 1TR6 on the ISDN side and
works fine
Hello all,
I just configure Bind 9 in our LAN to resolve the Asterisk name
sip.bussines.com for our phones.
I want that when a local extensión calls to another local extension, the
phone shows Extension@DNS name instead of Extension@ip address
like now happens.
In all my phones I configure
We have a problem in authenticating with a SIP server running PortaSIP.
first, my exten.conf says:
exten = _396262X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _39064040.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
and sip.conf:
register=390645416983:[EMAIL PROTECTED]/390645416983
[to-uni]
Hi!
I just bought an ISA phonejack and now I'm having some kind of problems
using it.
My system:
- Debian Woody
- P-II/333
- 192mb memory
- Kernel 2.6.5
- Asterisk 1.0 (installed as Woody-backport)
- slightly modified ixj-module
- ISA phonejack
- Internet via ADSL (768mBit downlink)
I connected
Hello,
I am having problems getting two xlite clients to communicate through
asterisk. I am getting an error message:
chan_sip.c:2753 process_sdp: No compatible codecs.
I have enabled all possible codecs in xlite (Menu - Advanced system
settings -Codec settings) and have added the appropriate
Hello,
I must admit I live in perpetual fear of forgetting to switch of html or rtf
formatting (useful for work) and top posting. I can understand the issue
with the former but can see absolutely no reason why top posting is such a
problem. In fact I'd far prefer it. I get to my e-mail
On Thu, 2004-11-11 at 13:18 +0100, Frank Sautter wrote:
well i have an icon diva quadbri card and i already tried uploaded the
1TR6 firmware, which seems to work so far.
the problem is, that the capi module and therefore chan_capi do not load
correctly.
[snip]
If you could provide some info
Joe Greco schrieb:
This turns out to be a basic netiquette issue for all the people who have
joined the 'net.
Oh yeah ... In every usenet group I'm joining this is an issue because
some people don't follow this rule - some other didn't.
I personally really hate top posting or ToFu as we say in
Here's a thought, anyone have ideas on how you could take registrations
from SIP/IAX users and run an AGI command using Asterisk? My goal would
be to enter the user/IP (after user reg's) into a MySQL database then
have other asterisk servers read from the same db. This would be for the
sake of
You could try adding the line insecure=very to the relevant section of
the sip.conf this would force asterisk to only validate the IP address
and not the user name (possibly but it is woth a shot)
Jason
On Mon, 8 Nov 2004 10:28:03 -0800, Randy Bush [EMAIL PROTECTED] wrote:
You could maybe
George Gardiner wrote:
I must admit I live in perpetual fear of forgetting to switch of html or rtf formatting (useful for work) and top posting.
A: Because otherwise we don't understand what you're replying to.
Q: Why top posting is so frowned upon?
Cheers,
Gilad
--
Gilad Ben-Yossef [EMAIL
Can * support a box with multiple nic cards correctly?
Background: small isp operation in the US has a rather large wireless
network covering multiple counties. The wireless net is an isolated
network using private IP's and nat'ing (via Cisco 7206). Their dsl
customers are on another isolated
If someone provides me with an answer to a question or provides information
to enhance my asterisk system, I don't care if they top-post or bottom-post.
Marv
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On Mon, 8 Nov 2004 20:19:42 -1000, Richard [EMAIL PROTECTED] wrote:
I have a question here. If both companies use 200 as their extension, how
can * tell which context a received sip call uses?
The received sip call will be placed in the context specified buy its
definintion in sip.conf
Jason
If someone provides me with an answer to a question or provides information
to enhance my asterisk system, I don't care if they top-post or bottom-post.
That could well be fine, but things rapidly get confusing as it moves from
providing a single answer to a simple question to having an
It's no issue to use more than one nic.
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Thursday, November 11, 2004 7:29 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] Multiple NIC's on * box?
Can * support a box with
Hi,
I Looked through tons of pages sofar no luck. Hopefully some one could
tell me the directions or relevant commands for the following.
If I have an outbound call with a normal PSTN number from * to an other
* or IAX provider but that */provider is not reachable because of a
network congestion
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Chris Modesitt
Sent: Thursday, November 11,
2004 12:49 AM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] No
Inbound CallerID Name Has me Stumped.
My Telco swears that I have Caller
ID (Name and Number)
Have a newly installed * box (RH ES3, current cvs head) with a tdm04b
(4-port fxo) connected to four US CO Centrex lines. Inbound calls are
being handled correctly via entries shown below. However, outbound
4-digit calling (eg, sip phone dials 8125 or iax2 call dials 8125)
always receives a CO
You can even setup a single nic to have multiple IP addresses in linux...
Jesse
On Thu, 11 Nov 2004 07:28:30 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
Can * support a box with multiple nic cards correctly?
Background: small isp operation in the US has a rather large wireless
network
hi there,
How about changing the general conf in sip.
disallow=all
allow=ulaw
allow=alaw
allow=gsm
not just disallow=all
and take them out of the extentions conf.
To me since you have the same codecs allowed its kinda not needed in
my mind to specify it to that level. Maybe it will fix
On Wed, Nov 10, 2004 at 11:02:14PM +0100, Elmar Haneke wrote:
how to configure * to send an SMS to an mobile phone (Germany, D2).
In the outgoing directory I do playe an call-file:
Channel: CAPI/[MYMSN]:0106301722270333
http://www.voip-info.org/wiki-Asterisk+cmd+Sms
SMS with T-Com
MvB:
Is this possible in Asterisk
Yes.
and what should be the approach?
Read the Wiki ;-)
http://www.voip-info.org/wiki-Asterisk+cmd+dial
Look at the 'g' parameter.
Nick.
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- Unit came with SIP v1.0.0.34 Release code 0035-00-00 installed. No
CDROM shipped with the unit, and a quick look at www.aastra.com
and www.sayson.com sites didn't appear as though one can download
firmware upgrades. Not sure where one is supposed to get them.
There is a
Cool. I thought that I had seen a few people posting over the last
several months that inferred * tied itself to a specific interface,
but I must have misread those postings. Thanks.
You can even setup a single nic to have multiple IP addresses in linux...
Jesse
What is the easiest way to record all parties of a meetme conference into 1
sound file?
I tried using Monitor just before the MeetMe call and it gave me files for
each person.
THanks,
Matthew
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Try to mix them and you will get 1 file ...
On Thu, 2004-11-11 at 16:40, Matthew Boehm wrote:
What is the easiest way to record all parties of a meetme conference into 1
sound file?
I tried using Monitor just before the MeetMe call and it gave me files for
each person.
THanks,
Matthew
Just a quick FYI for the Aastra/Sayson 480i SIP phone
Just received one and now have it running with *.
- Unit came with SIP v1.0.0.34 Release code 0035-00-00 installed. No
CDROM shipped with the unit, and a quick look at www.aastra.com
and www.sayson.com sites didn't appear as
I know this is on the wiki, I just want to confirm so I don't blow up my
cisco phones. I've got several cisco 7940's all running using cisco
power cubes. However, my boss wants me to switch just a few over to
poe, but doesn't want to fork out the dough for a nice cisco poe switch,
or anybody
There is an autocreatepeer flag in the sip.conf
http://voip-info.org/wiki-Asterisk+sip+autocreatepeer
That allows calls to go through without having to register.
Race Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd
Karlsbakk
There is still a very big problem with this phone, the dial plan will
only allow you to dial 10 digits. For local numbers this is not a
problem, but you cannot dial long distance.
edit the dial plan in the cfg file
# The dial plan that the 480i phone should use
# Where,
# 0, 1, 2, 3, 4, 5, 6,
Christopher
http://www.voip-info.org/tiki-print.php?page=Cisco+POE
Its ALWAYS on the wiki :)
Good question, but the 7940 is NOT a proper 802.3af (POE) device. It
is a polarity problem, which can be fixed with a crimp tool.
With 1 minute of crimping I have seen them work with the DLINK
I know this is on the wiki, I just want to confirm so I don't blow up my
cisco phones. I've got several cisco 7940's all running using cisco
power cubes. However, my boss wants me to switch just a few over to
poe, but doesn't want to fork out the dough for a nice cisco poe switch,
or
Yes,
Look in the wiki for bindaddr
bindaddr = 0.0.0.0 :IP Address to bind to (listen on)
http://voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.con
f
Be careful with the bind address. I know I have been burned by not
getting it right. Asterisk answers on eth0 but I am
By far the best poe (price/performance) I have seen for Cisco poe (or
standard poe) is the Netgear FSM7326P.
http://www.cdw.com/shop/products/default.aspx?EDC=568864
It is a managed layer3 poe switch (24 port) with 2 gigabit ports also.
Works out of the box with Cisco and Snoms (it auto detects
What is the easiest way to record all parties of a meetme conference into 1
sound file?
The easiest way is to Originate a call from the manager interface from a
Local extension that is setup to record(see example below) for a flat amount
of time and have it call into the meetme room. It'll
Joe Greco wrote:
*snipped
There's no reason, other than sheer laziness, to top-post. Providing
useful information might lessen the offense somewhat :-), but does not
(IMO) make it somehow okay to do.
... JG
so if by chance there is a thread you are interested in that 3 other
TP'er were
Hello,
I have a brief question, how do you format the following line in the sip.conf
file, the # in it seems to throw it off, but I have no option but to keep it on
the password
register = 1999555:[EMAIL PROTECTED]
I tried escaping the #, but I still can't get it to work
Thanks
Doug
Joe Greco wrote:
There's no reason, other than sheer laziness, to top-post. Providing
useful information might lessen the offense somewhat :-), but does not
(IMO) make it somehow okay to do.
so if by chance there is a thread you are interested in that 3 other
TP'er were engaged in, you
Thank you very much for your response. I was
wondering if it would be ok for me to ask you a couple
of additional questions.
1. Do you think this woul work?
http://www.phonegeeks.com/patpanwit25p.html
2. If I use the 25 pair (Amphenol) for hooking up
analog phones, what ports on the ADIT
It's been awhile since I've played with X-Lite, but I think it
absolutely *has* to use the MD5 auth stuff.
Use md5secret rather than secret in sip.conf. You'll have to MD5 hash
your password... there's documentation on this in the Wiki.
-Chad
On Nov 10, 2004, at 9:25 AM, Ashling O'Driscoll
[snip]
It's somewhat amusing, but mostly annoying, to see people fighting this
fight still even after 10+ years on the Internet.
In my experience, there will always be 2 kinds of posters in email
lists/USENET:
1) The somewhat intelligent comprehensive types who understand inline
posting and
Found the setup docs to convert cisco to SIP phone.
setup tftp
downloaded version 7.3 from cisco, put in /tftpboot directory.
reset the phone.
looked at the /var/log/messages and found this:
Nov 11 16:35:21 snorkel in.tftpd[4465]: RRQ from 192.168.1.85 filename
OS79XX.TXT
Nov 11 16:35:21
OK, the line's now set to ETSI, still having probs.
Anyone got some working configs ?
Steve
--
NetTek Ltd Phone/Fax +44-(0)20 7483 2455
SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19
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According to:
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip3
The phone should request the OS79XX.txt file from the TFTP server, and
after that should download the new firmware, and it shouldn't request
the SEPcnf.xml file. Are you sure
Yes, you can do this, in fact I'm sure most of the people who use
asterisk do this. Check out
http://www.voip-info.org/tiki-index.php?page=Asterisk for more
information about how to set up SIP channels and users.
-Chris
On Tue, 09 Nov 2004 00:19:54 +0500, Adnan Ahmed [EMAIL PROTECTED] wrote:
Here are the files in the directory.
[EMAIL PROTECTED] tftpboot]# ls
cisco.P0S3-07-3-00.zip OS79XX.TXT P003-07-3-00.bin P003-07-3-00.sbn P0S3-07-3-00.loads P0S3-07-3-00.sb2 SEP000FF78DEBB2.cnf
[EMAIL PROTECTED] tftpboot]#
According to:
I just tried the tftp localhost and "get OS79XX.TXT"
it says access violation.
Here are the permissions of the files. any idea on why I'm getting
access violation?
drw-r--r-- 2 nobody nobody 4096 Nov 11 11:35 tftpboot
[EMAIL PROTECTED] tftpboot]#
[EMAIL PROTECTED] tftpboot]# ls -l
On Thu, 11 Nov 2004 09:33:29 +0200 (SAST), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
On Tue, 9 Nov 2004, Michael George wrote:
The only difference to my extensions.conf file is that if I have:
exten = s,2,DISA(no-password, disa)
-- Executing DISA(IAX2/[EMAIL PROTECTED]/6,
[Apologies if this is a repost, I needed to subscribe to post through
GMANE.]
I have a use case where I must not allow/respect or at least restrict
the SIP 302 Moved Temporarily message that many SIP UAs send when the
user enables Call Forwarding.
This is because some calls are personal to the
Try:
SetGlobalVar(OH323_OUTCODEC=g723.1)
Michael.
M. Ehsanul Karim wrote:
Hello,
What would be the outcodec value for g723.1 (6.3k). I have g723
support which works with SIP (not pass thru) , but when I use OH323 it
always
Unsupported ${OH323_OUTCODEC} value (G72316K3)!
I have enabled all g723
I've had problems using bind to bind to only my lan interface on eth1.
It has no problem when I specify 0.0.0.0 it binds to all.
On Thu, 11 Nov 2004 10:36:55 -0500, Race Vanderdecken
[EMAIL PROTECTED] wrote:
Yes,
Look in the wiki for bindaddr
bindaddr = 0.0.0.0 :IP Address to bind
On Wed, Nov 10, 2004 at 11:48:58PM -0700, Chris Modesitt wrote:
My Telco swears that I have Caller ID (Name and Number) being sent to me over
our PRI's (I have called them a half dozen times to confirm). My gut feeling
is that they are lying to me, this is why.
First I decided to Look
Hello,
I was trying to install astGUIclient following the SCRATCH INSTALLATION
document.
After I finished Step (6.1) -- creating the MySQL asterisk database and try to
do http://10.10.10.15/astguiclient/admin.php, it failed. The following are the
warning or error messages:
Any idea where is
Rich Adamson wrote:
Cool. I thought that I had seen a few people posting over the last
several months that inferred * tied itself to a specific interface,
but I must have misread those postings. Thanks.
I have a bunch of Asterisk systems using VLANs to reach multiple subnets
over a single physical
Are you waiting until the start of the second ring cycle before
answering the phone?
CLID information is sent in-band between the first and second ring
cycles. If you interrupt this process (by answering the phone before
transmission is complete), you will not receive the CLID information.
Race Vanderdecken wrote:
when looking into the sipfriends table (using mysql sipfriends from
asterisk cvs version -r v1-0), I see timestamp and ipaddr set to
0/NULL. When looking into the CDR, the user has dialled out recently.
Also sip show peer xxx shows no data.
How can this be true?
A
I was trying to install astGUIclient following the SCRATCH INSTALLATION
document.
After I finished Step (6.1) -- creating the MySQL asterisk database and
try to
do http://10.10.10.15/astguiclient/admin.php, it failed. The following are
the
warning or error messages:
Any idea where is the problem?
Hi everybody,
Anybody could give me a little hint to apply the patch described below and
how to enable sfftobmp ? reading the post below, fax.php seems to be used to
mail the result but was not able to find it, do I have to write it ?
Thanks in advance,
jl
-Message d'origine-
De : [EMAIL
Just a reminder, if you are using the stock fedora kernel I'd recommend
rebuilding it without preemption turned off as I've experience kernel
panics from the zaptel driver. Digium tech support agrees (or at least
did a few weeks ago) that is was a problem.
Adam
Sean Kennedy wrote:
Got it,
The same happened to me on an old RH9
It´s a permission stuff..
Check mysql permissions for root, cron.. Also check passwords (you can
connect to mysql without password). You can edit dbconnect.php to use
another user (ex: root)
Guido Rebert
Network Manager
GrupoPyD - +54 11 4800
Anyone have an example dialplan string as to what is valid for
these phones. Their admin manual doesn't cover it.
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To UNSUBSCRIBE or update
Hello,
Maybe someone here can help me.
I am looking for VoIP software ( client )
on my Palm Tungsten. So I can make
use of my Palm and Asterisk server.
Thank you for help.
Bartosz
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Hi all,
i'm experiencing a problem using a Digi DataFire Micro V ISDN card.
I can't dial out nor recieve a call.
*CLI dial
Nov 11 18:18:13 NOTICE[-151061888]: channel.c:284 ast_alloc_uniqueid:
uid = asterisk-2806-1100193493.0
-- Executing Dial(OSS/dsp, Zap/g1/||trT)
XTEN http://www.xten.com the same people that make x-lite make a softphone
for handhelds. I use it on my handheld with pocket pc 2003.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Jozwiak
Sent: Thursday, November 11, 2004 12:15 PM
To: [EMAIL
XTEN http://www.xten.com the same people that make x-lite make a softphone
for handhelds. I use it on my handheld with pocket pc 2003.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Jozwiak
Sent: Thursday, November 11, 2004 12:15 PM
To:
What is the purpose of NoOp (no operation) if it does nothing?
among other things, it logs, so you can see a context being
entered. e.g.
[ext-foo]
exten = _X.,1,NoOp(ext-foo cid=${CALLERIDNUM})
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Matt Schulte wrote:
Here's a thought, anyone have ideas on how you could take registrations
from SIP/IAX users and run an AGI command using Asterisk? My goal would
be to enter the user/IP (after user reg's) into a MySQL database then
have other asterisk servers read from the same db. This would be
Steave,
OK, so they made changes to register string. I never had user number in my
register string. It was always;
register=1408215:[EMAIL PROTECTED] It worked that way for
about 11 months.
anyway when I included the user number, it started sending me invite
messages again.
Thnkyou for this
Found the setup docs to convert cisco to SIP phone.
setup tftp
downloaded version 7.3 from cisco, put in /tftpboot directory.
reset the phone.
looked at the /var/log/messages and found this:
Nov 11 16:35:21 snorkel in.tftpd[4465]: RRQ from 192.168.1.85 filename
OS79XX.TXT
Nov 11
Try here.. http://www.vliusa.com/prof_personal/index.php
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Jozwiak
Sent: Thursday, November 11, 2004 12:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Please clarify:
Fedore Core - build with preemption off or preemption on ?
The way you worded it, it's almost as if you're suggesting it with it
turned on?
Thanks!
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Adam Fineberg
Sent:
Hi,
With the Patch, now I see following log notices every 13-14 seconds on my
console for each SIP provider.
Nov 10 22:52:06 NOTICE[1089948224]: chan_sip.c:4023 sip_reregister:--
Re-registration for [EMAIL PROTECTED]
Nov 10 22:52:06 NOTICE[1089948224]: chan_sip.c:6795 handle_response:
X-Lite works fine for me with plain text passwords. Unlike the stuff
below, though, I'm not using nat=yes.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chad Scott
Sent: Thursday, November 11, 2004 10:20 AM
To: Asterisk Users Mailing List -
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ismaelg
Sent: Thursday, November 11, 2004 6:46 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk DNS issue
Hello all,
I just configure Bind 9 in our LAN to resolve the Asterisk
name
Thanks Matt, I will give that a shot tonight and will let
you knowJ
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Hello
Can or is there somewhere a way to test my outgoing H323
I like to connect to a terminating server but I'm still getting hangups.
Phone is ringing on the othersite but my asterisk telling my no one
availble at this moment.
Like to test my H323 loutgoing line.
I't looks so stuppid if
Hi - I have a zaptel card with 4 modules - 2 fxs and two fxo. I have two
phone lines coming into my house.
For now I want an incoming call to ring a phone here, and then if no
answer to ring another number (by calling out on the other line) for 15
seconds... then if no answer send to voicemail. It
Matt, I am unable to check-out libpri-matt,
is there something special I need to do? Let me know and Thanks!
cvs server: cannot find module
`libpri-matt' - ignored
cvs [checkout aborted]: cannot expand
modules
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I bought a Wildcard TDM400P earlier this week.
I compiled the software from CVS and installed it. When ztcfg runs I get
the error:
ZT_CHANCONFIG failed on channel 1: No such device or address (6)
After checking /proc/pci I don't see the board. Why wouldn't the board be
showing up? Its in a
Well, from what I'm looking at here, it appears preemption is off by
default ( installed the sources, did make menuconfig.
*shrug*
Thanks again
Sean
Adam Fineberg wrote:
Just a reminder, if you are using the stock fedora kernel I'd
recommend rebuilding it without preemption turned off as I've
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