I wouldn't have asked if it didn't matter. All I want is a place where
I can find the regulations. I already know how to use the Playback
command.
On Mon, 2005-01-31 at 06:45 +, Tom Shoval wrote:
Tim Mattison wrote:
Good call.
For our American readers... does anyone know where I
If the problem is in mpg123 than way just not to replace it?
Here is one very good example how to do it:
http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it
but I think that the problem maybe is coming from the BRIstuffed * - the
patches and etc.
Lubo
-
AppRadius Project: Full RADIUS
Have the same problem with PRI,
Probably the problem is in asterisk
MusicOnHold feature.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remco
Barende
Sent: Monday, January 31, 2005 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
please start asterisk -vvvcg (so it creates a core file when it
segfaults), then run gdb /usr/sbin/asterisk corefile, hit
Enter a few times and run a backtrace using bt. Please email
the output. I doubt that it's bristuff bug, since many users
have already successfully upgraded.
best regards
Yes, I do think the segfault problem is in bristuff. I just noticed
another problem though!
If there is one ongoing conversation and a new call is coming in, as soon
as the new call is answered, the call that was going on goes dead on one
end (the other party can hear the person callingfrom
On Mon, 2005-01-31 at 16:51, jurgen wrote:
Hi Howard,
Which provider are you with? We're with Primus Business here in
Melbourne, and haven't had anything like what you're describing. For
reference, here's a snip of my zapata.conf:
Big T
[channels]
language=en
context=local
Has anyone tried Sipura products such as the 3000 in Japan?
Jason
Steven Critchfield wrote:
On Mon, 2005-01-31 at 12:53 +0900, Jason Frisch wrote:
Sorry for my ignorance, but what is J1? I actually hope to use Softbanks
fiber-based IPtel
service, but I believe they require VoIP TA so I guess
Thanks everyone for your help.
The code in the dialplan was ok.
I had to switch to CVS head and everything worked straight away.
Any clues on when this will be working in the stable release ?
Thanks
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Try
busydetect=no
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes
Sent: Monday, 31 January 2005 19:17
To: jurgen; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Zap channels in AU
jurgen wrote:
Thanks for the suggestion, but it's no good. It still times out after
10 seconds. It seems to be something in the Monitor application,
rather than anywhere else. I can playback a sound (like the monkeys,
or MOH) forever and ever without timing out. Monitoring kills itself
though.
On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote:
Is anyone having/had a problem with a TDM400P card hanging up on STD
outbound calls as soon as the called party answers.
I'm guessing that * is responding to the STD pips in some way.
I had the same problem (before I
Can Asterisk work as Instant Messaging
Proxy?
Is there anybody who can help me?!
Best regards from
Italy
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On Sun, 30 Jan 2005 21:40:01 +0100, Philipp von Klitzing wrote:
Hi there,
this is just a short note about one of the PA168x based phones out there
which I obtained as Giptel G100 (aka Siptronic ST-100): For some reason
this phone would refuse to register with Asterisk using SIP, but after
Hi,
Does Asterisk have a limit to how many NAT'ed SIP clients it supports behind a
single IP?
I have the weirdest problem ever. I have three SIP endpoints. SNOM phones, if
it matters. Their extensions are 200, 201 and 202. Apart from the
username/password, the sip entries in sip.conf all have
Guys, On a telsttra line you can have the STD pip's removed,
I recently did this on a few on my lines.
If you want the setting to ask telstra for, ask me off list and i'll
try and find it.
On Mon, 31 Jan 2005 16:51:38 +1100, jurgen wrote:
Hi Howard,
Which provider are you with? We're with
Thanks for the replies everyone
how do you expect to get the indication that you have a
callwaiting call?
The whole point is I don't want it.
The beep is a guard that hides the
caller-id fsk spill also. So you can't get
callwaiting-callerid and not have a beep.
I don't really need
witht he bigt just ask provisioningto have NOPIPS set on the required
lines.
simple really
On Mon, 31 Jan 2005 19:17:05 +1100, Howard Lowndes wrote:
On Mon, 2005-01-31 at 16:51, jurgen wrote:
Hi Howard,
Which provider are you with? We're with Primus Business here in
Melbourne, and haven't
Did anyone get anywhere with this thread? Any HP G4 series servers working?
On Wed, 26 Jan 2005 09:46:31 +1100, Eric Bishop [EMAIL PROTECTED] wrote:
Has anyone had any luck with this issue and new Asterisk/Zaptel
releases (1.05/1.04)? I am still searching for a solution and waiting
for that
Do you have a config sample on how to handle digital PPP sessions in Asterisk?
On Sat, 29 Jan 2005 15:16:51 +0100 (CET), Peter Svensson
[EMAIL PROTECTED] wrote:
On Sat, 29 Jan 2005, David Norton wrote:
Currently I only have 1 PRI which I am using for dial-in customers. The line
is
Hi all,
I have this weird problem.
I'm running asterisk 1.0.3 on Debian Sid (official debian package).
We have 2 fritz ISDN cards.
All is working great.
Till I called the bank. It rings one time and then gives me
the congestion tone.
Here is what I see on the CLI (phone nr obfuscated for
privacy
You just have to define the accountnumber into your sip.conf/iax.conf.
Of course, accountnumber = card number of areskicc!
Then the IVR application wont prompt anymore to enter the cardnumber and
ask directly to dial the destination numer.
Hope this is help,
/Areski
On Sat, 2005-01-29 at 05:23,
Hello, i got a question,
i need to create a group extension, to make calls to 6 sw phones, but i
need to know if asterisk can do help me to get a unique number and check
what extension has received less calls than the others, and pass the new
call. We got a call center and want to know if we can
Hi all,
I'm using asterisk 1.0.2 (the Debian Sarge package) with Cisco 7905G
phones (SIP firmware). I've defined a macro to do some custom CallerID
stuff for our 100-number ISDN range (so we can see what line they've
called).
What I'd like to do is have the phone display update (to the original
On Monday 31 January 2005 09:29, Tais M. Hansen wrote:
Hi,
Does Asterisk have a limit to how many NAT'ed SIP clients it supports
behind a single IP?
[...]
Theoretical limit is around 65536 clients.
B
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Hi,Edgar,
Config the agents.conf correctly and it will do what you want. For more
information, search it in the wiki please.
Regards.
David
http://www.iaxtalk.com
- Original Message -
From: Edgar de Leon [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
This is the output of gdb:
Reading symbols from /usr/lib/asterisk/modules/cdr_pgsql.so...done.
Loaded symbols for /usr/lib/asterisk/modules/cdr_pgsql.so
Reading symbols from /usr/lib64/libpq.so.3...done.
Loaded symbols for /usr/lib64/libpq.so.3
Reading symbols from /lib64/libcrypt.so.1...done.
Edgar de Leon wrote:
Hello, i got a question,
i need to create a group extension, to make calls to 6 sw phones, but i
need to know if asterisk can do help me to get a unique number and check
what extension has received less calls than the others, and pass the new
call. We got a call center and
On Sun, 2005-01-30 at 03:11 -0600, Steven Critchfield wrote:
On Sun, 2005-01-30 at 12:31 +0330, Paradise Dove wrote:
hi,
just got an strange crash, and don't know what could cause this type of
crashs
- hardware failure
- memory
- cpu
?
i have 1xTE405P installed with
Hello,
I need some clarification on TDM400P.
While browsing the Digium site I see that the TDM400P wildcard
is being used as base card for other TDM series cards.
I wanted to know what basically the TDM400P card offers.
If you see the specs it says :
The Wildcard TDM400P is a
Hi
all,
I have trouble
getting my setup configured properly.
I have a Eicon|DIVA
Server BRI-2M/-2F card installed, using melware driver and following asterisk
wiki guidelines.
However whe I try to
dialup the number I get only silence and after a while
disconnection.
The following is
On Sun, 2005-01-30 at 14:02 +, Paul Tyreman wrote:
Hi,
This might not be a very popular question, but I was just wondering if
anyone have ever tried to run Asterisk on a Windows computer using Microsoft
Virtual Server
Hi All,
I have a question for you:
- "SIP doesn't work behind NAT very
well"
Do you agree with this sentence?
regards
César
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Can any give me or point me to a short and simple explanation of what HDLC is?
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Hi All,
I have a question for you:
- SIP doesn't work behind NAT very well
Do you agree with this sentence?
regards
César
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To
César Davi Ávila do Nascimento wrote:
Hi All,
I have a question for you:
- SIP doesn't work behind NAT very well
Do you agree with this sentence?
Depends on the NAT/firewall in question, you can also alleviate some of
these issues using STUN and sip proxing...
--
Best regards,
Duane
http://en.wikipedia.org/wiki/HDLC
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Eric Bishop
Gesendet: Montag, 31. Januar 2005 11:40
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [Asterisk-Users] HDLC for Dummies?
Can
http://fm.grandstream.com/gs/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert
Rozman
Sent: Wednesday, January 26, 2005 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk with Grandstream
On Mon, 31 Jan 2005, Eric Bishop wrote:
Do you have a config sample on how to handle digital PPP sessions in Asterisk?
No, but there may be examples in the wiki:
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+zapraspreview=3
http://www.digium.com/downloads/ppp.txt
chan_zap.c:3669: dereferencing pointer to incomplete type
chan_zap.c:3670: confused by earlier errors, bailing out
make[1]: *** [chan_zap.o] Error 2
make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.0.5/channels'
make: *** [subdirs] Error 1
[EMAIL PROTECTED] asterisk-1.0.5]#
It's not the
On Mon, 2005-01-31 at 15:30 +0500, [EMAIL PROTECTED] wrote:
Hello,
I need some clarification on TDM400P.
The TDM400P card by itself has no use. You purchase a mix of FXS and FXO
daughter cards (they are coloured Red and Green) which pug into four
available positions on the card. That
Hello,
I have been wanting to use digium x100p
to get started.
But it seems to have compatibility issues for different regions.
I am refering to the 600 ohm US pstn standard only.
I am in India so If I were not to use x100p card then
what card I need to go in for?
I also read
Leo:
Do you have a working MGCP call agent config? I've been struggling
with such a config for months and all my email queries have gone
unanswered. If you have such a config, and even better also have
SMDI support configured for/on Asterisk, I'd really appreciate a
copy.
Thanks,Steve
Leo Ann
So this leads me to believe there's some kind of limit per IP on
NAT'ed SIP clients.
Can anybody shed some light on this?
It sounds like a nat box issue and probably related to port mapping.
I've seen the same kind of issue with multiple vpn clients trying to
pass through a single nat box.
i need to create a group extension, to make calls to 6 sw
phones, but i
need to know if asterisk can do help me to get a unique
number and check
what extension has received less calls than the others, and
pass the new
call. We got a call center and want to know if we can distribute
Hi,
Does Asterisk have a limit to how many NAT'ed SIP clients it supports
behind a single IP?
[...]
Theoretical limit is around 65536 clients.
But the practical limit is something far less.
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Nathan
If you want more specific information for AUS, drop me a direct mail. My
Sipura 3000 passes the PSTN call (on hook) to the asterisk box and also the
CLIDNUM.
My only problem is that the asterisk box then sends the caller-id to the
handset connected to the sipura, I can get the username
Hi,
Enable the driver tracing (see wrapTrace* and libTrace* in oh323.conf),
re-run and send me the output file.
Michael.
Tola Ogunsan wrote:
Hi Michael and Everyone
I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm
getting this error
reason 24 (Call ended with Q.931 cause)
Hello,
I need some clarification on TDM400P.
While browsing the Digium site I see that the TDM400P wildcard
is being used as base card for other TDM series cards.
I wanted to know what basically the TDM400P card offers.
If you see the specs it says :
The Wildcard TDM400P is a
I have a question for you:
- SIP doesn't work behind NAT very well
Do you agree with this sentence?
Depends. Asterisk behind a nat box tends to be an implementation
problem limited by the knowledge of the person doing the implementation
and somewhat by the functionality implemented within
Hi,
I use a Cologne Chip HFC-8S card with chan_capi. (TE mode only)
So I've set up mISDN with CAPI and it's working just fine for'normal'
calls. But I do need more. Namely Hold, Retrieve and 3PTY which are not
supported yet in the mISDN implementation of CAPI, whereas it is in the AVM
Fritz
Thanks a lot!
Regards
César
- Original Message -
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, January 31, 2005 9:18 AM
Subject: Re: [Asterisk-Users] SIP x NAT
I have a question for
Yups... at least via FWD it is still
working.
Rene Kluwen
Chimit
- Original Message -
From:
Liaan vd Merwe
To: asterisk-users@lists.digium.com
Sent: Friday, January 28, 2005 4:48
PM
Subject: [Asterisk-Users] Fwd and
Tollfree
Hallo all
do any of you
szj wrote:
Hi, all:
I am a newbie to the asterisk and its architecture. :(
After reading some help in the tarball of Asterisk, I am
still in the mess. So I want to know where I can find a
detailed explanation of the Asterisk which including the
Architecture, Install, Configure, usage example
Michiel van Baak wrote:
All is working great.
Till I called the bank. It rings one time and then gives me
the congestion tone.
Here is what I see on the CLI (phone nr obfuscated for
privacy reasons):
-- Executing Dial(SCCP/michiel-0004, Modem/g1:xx|50|Ttr) in new stack
If you want
[EMAIL PROTECTED] wrote:
Hello,
I need some clarification on TDM400P.
While browsing the Digium site I see that the TDM400P wildcard
is being used as base card for other TDM series cards.
I wanted to know what basically the TDM400P card offers.
If you see the specs it says :
The
I have a question for you:
- SIP doesn't work behind NAT very well
Do you agree with this sentence?
Complete and utter crap (if you assume a few things).
SIP w/NAT works just fine if:
Asterisk itself is not behind NAT
You do not want to use SIP reinvites
You use some form of NAT Keepalive*
You must set trunk=yes in the context of the relevant provider. Not all
providers support it. The benefit of trunking grows exponentially with
the number of calls in progress.
-mark
On Jan 31, 2005, at 2:24 AM, Spencer Nassar wrote:
The test results that Philipp pointed out show some protocol
Rich Adamson wrote:
For example, the first sip session will use udp 5060, but on weird
nat boxes the second sip session will get mapped to udp 5061 (or
something like that), and obviously * isn't listening on that port.
The port that shows up in sip show peers is the remote SOURCE port
and
memtest86 is a nice tool and if you go to their site(http://memtest86.com),
they have an ISO bootable image there also.
Lyle
- Original Message -
From: Adam Goryachev [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
Tim Mattison wrote:
Good call.
For our American readers... does anyone know where I can obtain a list
of states/counties and their regulations in regards to call recording?
I would think the Public Utilities Commission for each state, but that's
just a guess.
A quick tickle of google came up
On 07:42, Mon 31 Jan 05, Eric Wieling wrote:
Michiel van Baak wrote:
All is working great.
Till I called the bank. It rings one time and then gives me
the congestion tone.
Here is what I see on the CLI (phone nr obfuscated for
privacy reasons):
-- Executing
I have one more question that I can't seem to get straight, The ZAP channel
phone, I can't dial any other extentions from it, I just get a fast busy.
Same if I dial 9 to use the outside trunk. It works great from the SIP soft
phone, but I can't seem to get the FXS phone to behave.
In your
asterisk -cv
Jan 31 18:03:20 WARNING[13145]: cdr_addon_mysql.c:264
my_load_module: Unable to load config for mysql CDR's:
cdr_mysql.conf
[app_addon_sql_mysql.so] = (Simple Mysql Interface)
[pbx_dundi.so]Jan 31 18:03:20 WARNING[13145]:
This is super easy to do. All you need to do is to put that announcement in
a MP3 and set a musiconhold class for that incoming Zap channel. Then
basically when ever that PSTN number rings, Asterisk will play the MP3
stream Your call may be monitored or recorded, please hangup if you do not
That URL has been locked down for resellers and vendors only for a couple of
days now. Pity, one of the good things about the Grandstream was their
freely available firmwares. Oh well, time to find another phone - the
Sipura 841 is looking interesting.
Craig
- Original Message -
From:
Even without any options I get the same result:
-- Executing Dial(SCCP/michiel-000b, Modem/g1:xx|50) in new
stack
-- Modem[i4l]/ttyI3 is busy
-- Hungup 'Modem[i4l]/ttyI3'
== Everyone is busy/congested at this time
When I call my cell the second after that all works fine.
The 2 ISDN
What you want is impossible!
How can you expect Asterisk to play a message to the caller without
answering the phone?
On Mon, 31 Jan 2005, Stefan Gofferje wrote:
Hi folks,
is there a chance to play an announcement to the calling party AFTER the
called party has picked up the receiver and
Does TDM400P wildcard has FXO and FXS ?
This card as 4 ports, and on each port you can put an FXS or FXO
module. So you can make any combination : 2 FXO and 2 FXS, 4 FXS, 4
FXO, etc
HTH
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-Original Message-
From: David Liu [mailto:[EMAIL PROTECTED]
Sent: 31 January 2005 14:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Announcement to caller when
called party haspicked up - without initial Answer()?
This is super
Hello Stefan,
Am Mo, den 31.01.2005 schrieb Stefan Gofferje um 15:09:
is there a chance to play an announcement to the calling party AFTER the
called party has picked up the receiver and WITHOUT asterisk answering
the call?
I would try the M option in the Dial-Command. See
Remco Barende wrote:
What you want is impossible!
How can you expect Asterisk to play a message to the caller without
answering the phone?
One-way audio before answer is a pretty standard telco feature with
PRI service in some parts of the world.
___
On January 31, 2005 08:57 am, Mark Eissler wrote:
You must set trunk=yes in the context of the relevant provider. Not all
providers support it. The benefit of trunking grows exponentially with
the number of calls in progress.
Isn't it just a linear savings?
1 call: UDP overhead + voice data
2
On Mon, 31 Jan 2005, Remco Barende wrote:
What you want is impossible!
How can you expect Asterisk to play a message to the caller without
answering the phone?
It can be done on isdn connection and over VoIP links as well. The reverse
audio path is (can be) opened before the answer. The
Hi
All,
I'm running Asterisk
on the following
vendor_id : GenuineIntelmodel
name : Celeron (Coppermine)cpu
MHz : 668.202cache
size : 128 KB
with 192 MB Ram
Audio coming from
Asterisk (the demo ) is excellent when using a SIP phone on the LAN to
Asterisk,
and when dialling in
from
On Mon, 31 Jan 2005, Alex Barnes wrote:
Also I was under the impression that in Europe calls are charged as soon
as you start ringing and not on pickup (this may be out of date as its
been a while since my school skiing trip ;-P )
Not in all countries at least. Sweden has always had the
On Monday 31 January 2005 14:28, Craig Guy wrote:
That URL has been locked down for resellers and vendors only for a couple
of days now. Pity, one of the good things about the Grandstream was their
freely available firmwares. Oh well, time to find another phone - the
Sipura 841 is looking
From what I have read and understood, the
On Mon, 31 Jan 2005, el Flynn wrote:
Jon Gabrielson wrote:
When there are no zap channels available, I signal congestion
using the following:
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Playtones(congestion)
exten =
Isn't it just a linear savings?
1 call: UDP overhead + voice data
2 calls: UDP overhead + voice data + voice data
3 calls: UDP overhead + 3xvoice data
etc...
without trunking the UDP overhead is repeated for each voice call
I know nothing about the IAX protocol but I wont let that
Remco Barende wrote:
Hi list!
I'm still trying to figure out about the groups in asterisk.
If I understand correctly, if you assign a certain group number and
you assign the same call group number to a sip device the device will
reing even though you did not specifically specify it in
Remco Barende ha scritto:
Hi list!
I'm still trying to figure out about the groups in asterisk.
If I understand correctly, if you assign a certain group number and
you assign the same call group number to a sip device the device will
reing even though you did not specifically specify it in
On Monday 31 January 2005 14:37, Alex Barnes wrote:
[...]
Also I was under the impression that in Europe calls are charged as soon
as you start ringing and not on pickup (this may be out of date as its
been a while since my school skiing trip ;-P )
I'm not sure that has ever been the case.
B
memtest86 is a nice tool and if you go to their site(http://memtest86.com),
they have an ISO bootable image there also.
Knoppix also can be used to test memory
On the boot prompt just type memtest and it will start the test
HTH
___
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Peter Svensson wrote:
Dial() application will answer the incoming line once it is ready to
bridge the two calls together. If nothing else then one can always modify
the Dial() application to play a specific sound just prior to sending the
answer. I have not checked if there already is a generic
dean collins wrote:
Im using tftp server that automatically loads on each reboot, for some
reason the last 2 files fail to load each time. (and I think this has
always been the case)
Aborted 192.168.16.32C:\Program Files\TFTP
Desktop\1.0.5.18\cfg000b82005c24
My apologies - Dan
On Mon, 31 Jan 2005, Dan Adams wrote:
From what I have read and understood, the
On Mon, 31 Jan 2005, el Flynn wrote:
Jon Gabrielson wrote:
When there are no zap channels available, I signal congestion
using the following:
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
If I understand correctly, if you assign a certain group number and you
assign the same call group number to a sip device the device will reing
even though you did not specifically specify it in extension.conf?
Can I use callgroups in such a setup, any config examples?
Which isdn channel are
Hi, Id like to have asterisk play a sound file while
a caller is waiting to be connected to an extension. I tried using music on
hold, but that seems to run in a loop, not playing from the beginning for each
caller.
Are there any other options? It doesnt have to be an
MP3 file. I tried
How do you want to play something on the line without answering it first ?
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I'm using * 1.0.3 on Gentoo 2004.3, zaprtc from bri-stuff for timing.
When I put a caller on hold, the volume of the hold music in the
callers ear is extremely loud. I'm using the default entry from the
musiconhold.conf:
default = quietmp3:/var/lib/asterisk/mohmp3
Volumes with a called or
still in the mess. So I want to know where I can find a
detailed explanation of the Asterisk which including the
Architecture, Install, Configure, usage example document.
The answers to the questions you've been asking are probably here:
Starter articles:
I'll agree with that sentence. There are many times when even STUN and so on
isn't going to help. In Guatemala, a lot of people end up with private IPs,
behind two NATs, etc. I've seen them aggressively timeout connections, limit
the range of ports available for NAT (to a ridiculously low number),
Am I doing something wrong here? GOt a SPC-841 the other day and have it
registering properly. Can place and recive calls as expected but when on
the phone, a second call is immediately dumped to busy voicemail. Does
this thing not support call-waiting? Or, have I just got my configs
wrong?
On Mon, 31 Jan 2005 06:12:53 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
If you incorrectly connect a fxs module to a pstn line, you will
likely blow the module making it useless.
Wow, i think it shoud be more fool-proof ;) Lucky I tried first with
an analog phone in my TDM11B...
Julian.
Gents,
I've recently built a couple of Asterisk boxes and want to migrate
away from CallManager to Asterisk.
On my Asterisk box I have about 8 Grandstream BT101s and a
Cisco 7905G in SIP mode, on my CallManager I have about 10
x 30VIP, 2 x 7940 and a 7960.
I've built Asterisk version 1.0.5 along
Apparently I have had a few calls show up in my logs as something odd
happening. Apparently at a certain spot the wrong number of digits are
being presented, but I am not sure why that is. That is what I am trying
to figure out. I was curious, does anyone know of a wiki page that
outlines the
Nope the files are there.
Extracted the entire zip file into the same folder.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen R.
Besch
Sent: Monday, January 31, 2005 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
I am noticing a problem with the RT31P2-NA when it loses internet. Has
anyone
experienced problems where it does not reconnect to asterisk and obtain
its dialtone
again?
Brian Fertig
Planet Telecom, Inc.
___
Asterisk-Users mailing list
Hi,
Is it possible to send calls that are forwarded from a Cisco 7900 phone
using the Call Forward All feature out using a different service
provider or group like another SIP trunk? I don't want to tie up our
incoming lines that are ZAP so I was thinking about getting a secondary
service for
Stefan if I understood what you need, maybe this works...
MSG_FILE=/var/spool/asterisk/
exten = s,1,Dial(SIP/MyPhone|60|M(playmessage^${MSG_FILE}))
[macro-playmessage]
exten = s,1,Wait(0.5)
exten = s,2,Playback(${ARG1})
exten = s,3,SetVar(MACRO_RESULT=CONTINUE)
I didn´t try it but I think
On Mon, 2005-01-31 at 09:37 +, Alex Barnes wrote:
Thanks for the replies everyone
how do you expect to get the indication that you have a
callwaiting call?
The whole point is I don't want it.
The beep is a guard that hides the
caller-id fsk spill also. So you can't get
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