Hi,
Does anyone have sample configuration files for a Cisco 7905G to use
with SIP/Asterisk ?
I'm on Firmware 3-08-12 - is there a better release to run ?
/S
--
_
-- Bandwidth and Colocation Provided by
Hi Asterisk Users,
Sorry if this is off Topic for this list.
But does anyone have a full XML config file for the SPA-3000, the PAP2 and
the SPA-941.
Or alternatively a way to convert the field names on the web pages to the
corresponding XML filed names.
Thanks
/S
Hi,
I'm looking for a great tech support person to take over the admin of
our asterisk system. If you are a networking person as well, with some
experience in firewalls and desktop support even better. The system is a
multi-group system with IVR, Follow-me dialing, voicemail, and
Hi,
I have a setup where I have multiple lines to the same provider - in
this case broadvoice.
I have created the entries in sip.conf for both accounts - and
independently they work fine. When they both are in use, incomming
calls are placed to the one that's the last in the sip.conf file.
.
/Soren
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messing up Zaptel on 2.4.
I have edited:
pciradio.c tor2.c torisa.c wcfxo.c wct1xxp.c wct4xxp.c wctdm.c
wcte11xp.c zaptel.c ztdummy.c ztdynamic.c
and changed:
#include linux/moduleparam.h
to:
#ifdef LINUX26
#include linux/moduleparam.h
#endif
and now it compiles on 2.4...
/Soren
Following is sharelessly copied from one of the newsgroups I read on
grc.com..
/Soren
NIST issues recommendations for secure VOIP
http://www.gcn.com/vol1_no1/daily-updates/34747-1.html
http://csrc.nist.gov/publications/nistpubs/800-58/SP800-58-final.pdf
exten = i,1,Wait(1)
exten = i,2,Answer
exten = i,3,Playback(invalid)
exten = i,4,Hangup
;;;end of extensions.conf
More here: http://www.voip-info.org/wiki-Asterisk+i+extension
/Soren
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already been upgraded to SIP 7. Now the box
is ready but we don't know what the next step is!!
Any help is appreciated.
http://your_server/maint
/Soren
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device/context by IP address and not Name...
It works for me but it is apparent to me that the h323 stack in the phone is
pure crap.. :-)
/Soren
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Andrew Kohlsmith wrote:
On December 13, 2004 03:10 am, Soren Rathje wrote:
wait_just_a_bit(HZ/10);
I didn't want to wait inside the driver, likely a place where
interrupts are disabled...
Well, nobody claimed it was ready for production.. :-) I'm usually OK for
POC code, but don't
Greg - Cirelle Enterprises wrote:
it appears the cvs for zaptel as of 12/13/04 am is missing
at least 1 file -- wcfxs.c
It was renamed to wctdm.c around Nov. 6. 2004
/Soren
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*/
else {
wctdm_setreg(wc, card, 72 , reg72 0x3F);
wait_just_a_bit(HZ/10);
wctdm_setreg(wc, card, 72 , reg72 ^ 0x40);
}
I just haven't found the place to put it..
/Soren
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Thorben G. Jensen wrote:
How do I Playback the Mailbox Owners Name?
Ex.: I want a message saying I am sorry but + Mailbox Owner Name +
has gone to lunch
Extension 999 in voicemail context internal
exten = 999,1,SetVar(VM_CONTEXT=internal)
exten = 999,2,Playback(im-sorry)
exten =
Soren Rathje wrote:
Specs for Si3210 (TDM400P FXS Module) says on page 93:
---
Register 72. On-Hook Line Voltage
Bit 6 VSGN On-Hook Line Voltage.
The value written to this bit sets the on-hook line voltage polarity
(VTIPVRING).
0 = VTIPVRING is positive
1 = VTIPVRING is negative
Module) says on page 93:
---
Register 72. On-Hook Line Voltage
Bit 6 VSGN On-Hook Line Voltage.
The value written to this bit sets the on-hook line voltage polarity
(VTIPVRING).
0 = VTIPVRING is positive
1 = VTIPVRING is negative
---
I wonder if this can control Pol-Rev's.. ?
/Soren
stuff like that but
it looks like the thing you want. I'm sure there are other alternatives out
there that will do the same only differently...
/Soren
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://www.voip-info.org/wiki-Asterisk+x100p+echotraining) you can train the
ears of Asterisk.
/Soren
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termination
mode (DC 600 ohms) and you have gained nothing.
/Soren
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.
Alternatively the TDM400P is using a global chipset -- WITH SUPPORT FROM
DIGIUM.
Follow your conscience :-)
/Soren
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Adnan Ahmed wrote:
hi,
I am not registered my SIP Phone with Asterisk i spend almost one
day but find no luck my configs are.
Please post console log with errormessage..
My guess is the host=192.168.10.195 definition and the use of
context=sip not matching the dialplan.
/Soren
Michael Vogel wrote:
Soren Rathje schrieb:
Note: The Wildcard X100P/X101P only have FCC approval.
What does that mean for me? Is it illegal to use it in germany or do
they don't work in germany?
They will *probably* work in Germany..
Check with T if they allow non-CE approved equipment
- old picture, new card,
different chipset.
http://cgi.ebay.de/ws/eBayISAPI.dll?ViewItemcategory=8057item=6724669731rd=1
http://cgi.ebay.de/ws/eBayISAPI.dll?ViewItemcategory=8057item=6723748329rd=1
/Soren
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with a Gatekeeper to see if that changes
anything..
Powersupply
Type: FW 6798 (Made in Germany)
Input: 230V~/50Hz/73mA/16,8VA
Output: 19V~/600mA/11,4VA
Plug (phone): RJ12 pin 1+6
/Soren
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asks may be a fool for five minutes. But he who does not ask
remains a fool forever.
On the other hand... This may be more appropriate...
Accept that some days you're the pigeon, and some days you're the statue.
Scott Adams.
:-)
/Soren
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)
Take a look at zonedata.c, it should be obvious to you as you already worked
on the indications.conf file.
Once you recompile Zaptel and everything works you can submit the changes to
bugs.digium.com to make the internationalisation more complete. :-)
/Soren
Thomas Andrews wrote:
On Sun, Nov 14, 2004 at 11:46:15AM +0100, Soren Rathje wrote:
Can you post your actual configuration ?
/etc/zaptel.conf
fxols=1 #S100U
fxsls=2 #X100P
loadzone = us
defaultzone=us
Looks fine allthough the comments are wrong :-)
/etc/asterisk/zapata.conf
terminology.. :-)
/Soren
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Thomas Andrews wrote:
On Sun, Nov 14, 2004 at 02:22:18PM +0100, Soren Rathje wrote:
/etc/zaptel.conf
fxols=1 #S100U
fxsls=2 #X100P
loadzone = us
defaultzone=us
Looks fine allthough the comments are wrong :-)
Thanks Soren. I made all the changes you suggested, but do I have
Thomas Andrews wrote:
On Sun, Nov 14, 2004 at 03:01:28PM +0100, Soren Rathje wrote:
Hmm.. Does Asterisk load chan_zap ?
I believe so:
[chan_zap.so] = (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, FXO Kewlstart signalling
Thomas Andrews wrote:
On Sun, Nov 14, 2004 at 03:16:13PM +0100, Soren Rathje wrote:
Hang on... What line pair do you use on the phone; 1+4 or 2+3 ?? I
believe the correct pair to use should be 2+3.
It's the middle pair. I assume that's 2+3 on an RJ connector ?
Correct..
Just
insmod wctdm debug=1
/sbin/ztcfg
Now you can tail -f /var/log/messages and see hookstate. Already at this
point I get a dialtone on my FXS port.
/Soren
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Thomas Andrews wrote:
kernel: NO BATTERY on 1/2!
I don't like the look of that NO BATTERY message. What do you think
Soren ?
NO BATTERY applies to FXO ports and says that Span 1/Card 2 does not
receive power from the line, i.e. it is not plugged into the wall socket.
(if I read the source
Thomas Andrews wrote:
On Sun, Nov 14, 2004 at 04:36:21PM +0100, Soren Rathje wrote:
NO BATTERY applies to FXO ports and says that Span 1/Card 2 does
not receive power from the line, i.e. it is not plugged into the
wall socket. (if I read the source correctly)
ok. I connected it to the PABX
Loopstart (Default) (Slaves: 02)
2 channels configured.
I've used this link to set up the ports:
http://www.digium.com/index.php?menu=faq#Configuration_0
Can you post your actual configuration ?
/etc/zaptel.conf
/etc/asterisk/zapata.conf
/Soren
Communications product sheet (www.amigocom.com)
http://61.31.72.100/amigocom/products/ami_ia92_ie92.htm
/Soren
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'opermode=1'.
/Soren
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modules in
/etc/asterisk/modules.conf if you do not want to delete them.
/Soren
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Asterisk using:
/usr/sbin/asterisk -vv -g -dd -c
/QUOTE
/Soren
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can to
make a viable solution. If you are successfull in doing so please share with
others via the bugtracker.. :-)
Regards
Soren Rathje
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=9, there has been some development here. UK BT
CLIP and DTMF CLIP for the TDM400 (will not work for the X100P) is now
merged into one patch.
/Soren
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should air this on grc.com, some of the propellarheads there may have a
clue...
/Soren
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John Todd wrote:
At 10:09 PM +0200 on 8/10/04, Soren Rathje wrote:
John Todd wrote:
At 7:14 PM +0200 on 8/10/04, Soren Rathje wrote:
Gang,
[snip]
/Soren
It is the mark of an educated mind to be able to entertain a
thought without accepting it.
- Aristotle
Ok, so we moved
:
..
[spouse-factor]
exten = s,1,NoOp(${CALLERID})
[next-context]
..
No Wait() or Answer() so the line will never be answered but incoming callerid will be
in the log/cdr... :-)
/Soren
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is usually received after
the first ring and before the second ring.
/Soren
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David Cook wrote:
Quoting From: Soren Rathje [EMAIL PROTECTED]
No Wait() or Answer() so the line will never be answered but incoming
=
callerid will be in the log/cdr... :-)
/Soren
I think I just missed something very fundamental. You are saying that
the switch doesn't pickup the PSTN
John Todd wrote:
At 7:14 PM +0200 on 8/10/04, Soren Rathje wrote:
Gang,
[snip]
/Soren
It is the mark of an educated mind to be able to entertain a thought
without accepting it.
- Aristotle
Ok, so we moved here from *-dev, no problem... ;-)
VOIP Spam is actually pretty trivial
1 Record your unavailable message
2 Record your busy message
3 Record your name
4 Change your password
* Return to the main menu
/Soren
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, pick-up and continue). How long time before you see a hangup if you
leave the PSTN side on-hook after the call ??
-- Soren
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) and with a little tinkering
in your dialplan you can even activate/deactivate this from the Manager Station
Why make it harder than it really is ??? I believe this is exactly what they do when
programming your regular (old world) PBX systems...
-- Soren
Daniel Jimenez wrote:
Soren Rathje wrote:
Eh... Sort of like shadow lines ???
Remember that Dial(SIP/1 H323/1
ZAP/1,[timeout],[options],[URL]) will dial all 3 extensions
simultaneously (regardless of channel choice) and with a little
tinkering in your dialplan you can even activate
Jean-Yves Avenard wrote:
There's just what thing I can't figure out.
What is the action for s-.
It's the better safe than sorry option... :-)
Basically it's a wildcard option, anything beginning with s- will go there...
-- Soren
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PROTECTED];tag=0939785f3bc7641e
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0
What are your codec settings in sip.conf ??
Could you try (can be set at client level):
disallow=all
allow=ulaw
-- Soren
On Friday 09 July 2004 15:30, Soren Rathje wrote:
What are your codec settings in sip.conf ??
Could you try (can be set at client level):
disallow=all
allow=ulaw
codec's are set to allow all.
I can't see how this would help. I can talk fine from local client to remote
so
,Voicemail(b${ARG1})
exten = s-BUSY,2,Goto(default,s,1)
exten = s-ANSWER,1,Playback(demo-thanks)
exten = s-ANSWER,2,Hangup()
exten = s-CANCEL,1,Playback(demo-thanks)
exten = s-CANCEL,2,Hangup()
exten = s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
exten = a,2,Hangup()
-- Soren
RFC1918 with CIDR notation
localnet = 169.254.0.0/255.255.0.0 ; Zero conf local network
Also, I saw some fixes to RTP address binding in CVS today. Hard to tell really
without a trace..
-- Soren
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is firewalled and UDP 5060 and 1-2 is forwarded to internal address.
External SIP clients have nat=yes, I dont see any difference with local clients if
they have nat=yes or nat=no.
-- Soren
- Original Message -
From: Ryan Courtnage [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent
, something like this for
starters...
-- extract from extensions.conf --
[voip-h323]
;
; OH323 default context from oh323.conf
; Dial 0[number]
;
exten = _0.,1,NoOp,${CALLERID} - ${EXTEN}
exten = _0.,2,Dial(IAX2/demo:[EMAIL PROTECTED]/${EXTEN:1})
exten = _0.,3,Hangup()
-- Soren
- Original
.
-- Soren
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to backstep from one kernel
release to another without crashing everything..
-- Soren
- Original Message -
From: TC [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 24, 2004 3:18 AM
Subject: Re: -- [Asterisk-Users] Serious issues with current CVS?
YUP lots of total weridness i
- Original Message -
From: Carlos Medina [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, June 22, 2004 10:47 PM
Subject: [Asterisk-Users] Unify Incoming and Outgoing sound files
Hi, i have a call center which receives many calls at day. Those calls are stored in
a directory in
is that this also applies for other telco's, the question is if they are
willing to disclose the information or not.
-- Soren
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tell me more hints about what you are writing!
Thank you!
Roger.
Google is your friend...
http://www.google.com/search?q=site:www.voip-info.org+asterisk+bounty
finds..
http://www.voip-info.org/wiki-Asterisk+bounty+SS7
apparently worth USD 3,000
-- Soren
+0200
From: The Asterisk PBX asterisk(atsign)domain.com
To: Soren soren(atsign)domain.com
Subject: New VM (1) - 2:04 long in mailbox 100 from Joe User 12345678
Message-ID: Asterisk-1-100-2792(atsign)asterisk.domain.com
On your server you install/setup/configure sendmail and have it point to your
, pull the string tight so the
match/stick is flat against the coin. Now, pull one end of the string while holding
the coin Get it ??
Not that many devices can do this... And if the do, they cost really big money... :-)
-- Soren
- Original Message -
From: Didelot Loic [EMAIL PROTECTED
if we need to substitute our
* externip or can get away with our internal bindaddr
*/
Then again, I could be wrong.. ;-)
-- Soren
- Original Message -
From: Didelot Loic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 11, 2004 2:37 PM
Subject: [Asterisk-Users] phone calls
and powerful with extension of C/C++
scripts.
Anyone know of a similar free product ??
-- Soren
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demo mp3, you'll know what I mean..
-- Soren
- Original Message -
From: Thomas Gallaway [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, May 10, 2004 9:00 PM
Subject: Re: [Asterisk-Users] I love you!
[EMAIL PROTECTED] wrote:
lovely, :-)
Is it just me or where there allready
|halvfems|tusinde|ni|hundrede|ni|og|ni|og|halvfe
ms
(200...X)
to|millioner...X
-- Soren
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: Users Asterisk [EMAIL PROTECTED]
Sent: Monday, April 19, 2004 9:53 PM
Subject: [Asterisk-Users] One, två, tre, quatre, cinq
I saw in the current CVS version that a new parameter has been added to
app_dial.c...
The option string may contain zero or more of the following characters:\n
't' -- allow the called user transfer the calling user\n
'T' -- to allow the calling user to transfer the call.\n
'r'
Ok, it actually works fine here..
Asterisk CVS-03/06/04-14:35:21, Copyright (C) 1999-2004 Digium.
From extensions.conf:
[pstn-out-nat]
;
ignorepat = 0
; NOT USED
exten = _0XX0X,1,Congestion
; Local eight-digit dialing accessed through trunk interface
exten =
Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
HMM - This wont work :(
exten = 10,1,Dial(SIP/hha1,20,S(10))
exten = 10,2,VoiceMail,u10
exten = 10,102,VoiceMail,b10
When did you checkout your version of Asterisk from CVS ??
This feature was put into CVS on
, I'm about three weeks into my very first * installation (that sort of
works), so basically any info/tips/tricks/word of advice is accepted with
appreciation...
-- Soren
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on or not..
-- Soren
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would be:
a: check if client is registered and/or reachable, if not - return
unreachable
b: check if client is busy, if call-waiting not active - return busy
c: if call is rejected by client, return approriate message
d: if call is unanswered, return unavailable or busy with reference to
(b).
-- Soren
announce
exten = s,3,Goto(default,s,1) ; If they press #, return to
start
exten = s,102,Voicemail(b${ARG1}) ; If busy, send to voicemail
w/ busy announce
exten = s,103,Goto(default,s,1); If they press #, return to
start
*
/Soren
--
It is the mark
in ETS 300 659-1 and ETS 300 659-2
if anyone is interested, and can be found here:
http://www.secret100.nm.ru/ets300659.pdf and
http://www.secret100.nm.ru/ets3006590e02.pdf
(I know, weird links, Google found them for me)
Thanks
Soren
--
It is the mark of an educated mind to be able to entertain
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