Chris Albertson wrote:
Your log file almost looks like a bug in Asterisk doesn't it?
Why call poll() with a zero timeout while passing only one FD?
and then why do the read when there is no data?
Read the man pages for all the system calls
Take a look at the source chan_sip.c
/* Wait for
Hi,
-Original Message-
look for a T1 failover switch.
(cheap as dirt on ebay, mine was $7.50, - yes really, I got
the decimal in
the right spot, hard to find an empty rackmount box that is cheaper.)
Nice tool. Anyone know of an E1 equivalent ? :-))
Best regards,
Florian
Hi,
-Original Message-
OpenSS7 project mentions Asterisk also. I think project will
bring something
what we all really need - SS7 support for Asterisk
Take a look : www.openss7.org
Actually, from what I remember this initiative focusses at creating and
maintaining SS7 driver
yes, mysql compilation into vm2 is broken, see my patch at
http://bugs.digium.com/bug_view_page.php?bug_id=441
these patches fix the compilation problem of mysql-vm-routines
In voicemail.conf, however, there is no paramter to specify a port
(or socket), at least not from what I read here
There were some fixes posted the other day.
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Bagyenda
Sent: Tuesday, 28 October 2003 3:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] RE: Voicetronix OpenLine4
Hi Jorge,
The
If I receive a call it is fine
But if I make a call the voice level is not so good and the echo is
shocking.
Any ideas would be appreciated
Regards Mick
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Many thanks for this.
This is more proof of concept than mission critical. I will need to
pinch some channels and send them to SIP phones etc.
Am I correct in assuming that the hardware I suggested will do exactly
this, or even a E410 so I could connect a couple of PBXs in my workshop
for
This looks like the same case as the thread:
Problem ATA-711-723-Oh323-Asterisk that was posted back in August.
Although, given the nature of trampled memory problems it could be almost
anything.
The Oh323 library dies in a call to realloc. This is almost certainly
indicative of someone
On Tue, 2003-10-28 at 03:14, Stuart Mackintosh wrote:
Many thanks for this.
This is more proof of concept than mission critical. I will need to
pinch some channels and send them to SIP phones etc.
Am I correct in assuming that the hardware I suggested will do exactly
this, or even a E410
== Parsing '/etc/asterisk/adsi.conf':
Found -- Accepting call from '890003' to '185' on channel
27, span 1 -- Executing Answer("Zap/27-1", "") in new
stack -- Executing Record("Zap/27-1",
"soundexampless:mp3") in new stack -- Playing
'beep'WARNING[360468]: File translate.c, Line 128
On Tue, 2003-10-28 at 04:43, Alexandru Coseru wrote:
== Parsing '/etc/asterisk/adsi.conf': Found
-- Accepting call from '890003' to '185' on channel 27, span 1
-- Executing Answer(Zap/27-1, ) in new stack
-- Executing Record(Zap/27-1, soundexampless:mp3) in new stack
--
Brad:
Lucent 5ESS can treat a PRA (PRI) like a CCS7 or R2 route. This feature
allow to route any number, including numbers outside the DID range. I know
some class 5 switches (Lucent, Siemens, Ericsson, Nec, Alcatel and Italtel),
and only Lucent 5ESS and Siemens EWSD provides this feature (at
I didn't know it... excellent!
- Original Message -
From: Thorsten Lockert [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 6:36 PM
Subject: RE: [Asterisk-Users] Music Onhold Configuration
MPG123 is not included in Asterisk...
Download the package:
Hello all,
Apologies as not really an Asterisk question - QOS. I have been told to
implement VOIP correctly you need QOS implemented across the network as
a whole. What network switches support this?
Regards
Nick
___
Asterisk-Users mailing
Today's CVS gives me an error while
compiling:
chan_zap.c: In function
`zt_train_ec':chan_zap.c:1076: `ZT_ECHOTRAIN' undeclared (first use in this
function)chan_zap.c:1076: (Each undeclared identifier is reported only
oncechan_zap.c:1076: for each function it appears in.)make[1]: ***
cvs update your Zaptel code and reload the appropriate kernel module.
Jeremy McNamara
Bartosz Jozwiak wrote:
Today's CVS gives me an error while compiling:
chan_zap.c: In function `zt_train_ec':
chan_zap.c:1076: `ZT_ECHOTRAIN' undeclared (first use in this function)
chan_zap.c:1076: (Each
But I am not using Zaptel at all.
Still I have to install Zaptel ?
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 28, 2003 9:05 AM
Subject: Re: [Asterisk-Users] Today's CVS
cvs update your Zaptel code and reload the
Title: Message
I have
the same problem any one
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexandru
CoseruSent: Monday, October 27, 2003 1:10 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] SIP -
H323 Seg fault.
A very
Music-on-hold and X-Lite:
Ok, finally I understand what is going on! :-) My chopppy music-on-hold
problem with my three * boxes was caused by the silence setup in X-
Lite:
Menu -- Advanced Settings -- Audio Settings -- Silence Settings
With Transmit Silence = Yes everything is fine, however
Pretty much anything from Cisco or Foundry support QOS. Linux and BSD
support it as well.
-sb
-Original Message-
From: Nick Knight [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 28, 2003 6:52 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] QOS
Hello all,
Apologies as not
Close. Normally, at least in Qwest-land, third-party VM provider systems
dial
into the switch and give it a DN and a MWI on-or-off command. If the DN
is
serviced by that switch, it turns the message waiting indicator (stutter
dialtone, MW light or both) on or off. If the number is on a
The problem should be easy enough to solve for someone who knows the
internal guts. As a matter of fact, this is very important to resolve.
Asterisk behind firewall is trouble and that is known already. So I
decided to use the same linux box as firewall, meaning I need atleast
two NICs. I wonder
This was triggered by the lack of an mp3 encoder. Without a backtrace
there's no way to know it's fixed for sure, but if you cvs update it
should at least fail cleanly and if not please place a bug in the bug
tracker.
Mark
On Tue, 28 Oct 2003, Alexandru Coseru wrote:
== Parsing
Update your zaptel.
On Tue, 28 Oct 2003, Bartosz Jozwiak wrote:
Today's CVS gives me an error while compiling:
chan_zap.c: In function `zt_train_ec':
chan_zap.c:1076: `ZT_ECHOTRAIN' undeclared (first use in this function)
chan_zap.c:1076: (Each undeclared identifier is reported only once
I don't know if this helps, but I run 2 Nics; one for access to the
192.168 (private) domain, the other for the public domain. Under RH the
second card was autodetected, installed and became available in the net
setup, where I assigned IP addresses to the cards and off I went. No
troubles at
I'm just finishing the test of a solution where the Asterisk box acts as a
firewall between the outside world and the inside world, but uses only a
single network card. It uses the VLAN capabilities built into Redhat 9.0.
As a consequence, the switch to which it is connected needs to understand
...I don't know what Grandstream programmed
into their phones; someone might help here.
Grandstream RTP is on ports 1024-65535 with port 5004 used as the default.
Stephen R. Besch
___
Asterisk-Users
-Original Message-
From: Thorsten Lockert [mailto:[EMAIL PROTECTED]
I addition to this, all the releases that has been made to date (0.5.0
for
Asterisk,
for instance), have been tagged. This means you can check out
Asterisk
0.5.0 by just
doing cvs checkout -r v0-5-0 asterisk once
Thanks very much.
It is working now!
Bart
- Original Message -
From: Mark Spencer [EMAIL PROTECTED]
To: ASTERISK USERS [EMAIL PROTECTED]
Sent: Tuesday, October 28, 2003 10:39 AM
Subject: Re: [Asterisk-Users] Today's CVS
Update your zaptel.
On Tue, 28 Oct 2003, Bartosz Jozwiak
Hello,
I have problem with ringing
applecation.
When I dial from a router Cisco 2600 with H323 to
Asterisk IVR and I dial extension I do not hear the ringing tone but phone is
calling.
But when I dial to asterisk IVRwith SIP phone
and dialing extension I can hear ringing tone.
What can be
Title: Software FAX Modem--One Last Request For Help
Here's one last plea for help from the list before I give up on the totally cool software FAX modem concept in complete despair (for now, at least, until I have more time to dig into it)
I am simply not able to successfully receive a
So, the proper answer is that if you really want to
implement this PRI - SS7
- PRI message, you should really be talking to your nearest
CO Engineer or
Telco Enterprise Business Office where they handle this all
the time for
enterprise call center applications.
Hah. I've yet to
Hi *ers,
If anyone with the capability and more appropriately the time, fancies developing a
patch to provide sip debug ip_address capability with Asterisk I am sure they will
be eternally praised (c;
Rgds, Adam
* DISCLAIMER *
This message and any attachment are
thanks for explanation.
It does not solves this problem, but another one :)
best regards
hudecof
Olle E. Johansson wrote:
Philipp von Klitzing wrote:
You will probably have to use canreinvite=no in the UA definitions
in the SIP.conf for those two phones..
In your
I have installed G729 but I cannot make a outgoing
call with it.
SIP/dennis-2c23 is making progress passing it
to SIP/1010-8b60NOTICE[311316]: File channel.c, Line 1476
(ast_set_read_format): Unable to find a path from G729A to
ALAWNOTICE[311316]: File channel.c, Line 1446
Nick,
Apologies as not really an Asterisk question - QOS. I have been told to
implement VOIP correctly you need QOS implemented across the network as
a whole. What network switches support this?
That is a very safe statement to make.
In the real corporate world, QoS may not always be
Florian,
Why would you need another box? Couldn't you just provision one PRI card for
asterisk and the other for SS7?
Brad
Florian Overkamp wrote:
Actually, from what I remember this initiative focusses at creating and
maintaining SS7 driver support for the Zapata PRI cards, not integration
Hi,
I just updated my image from CVS, compiled and reinstalled it. Now
whenever I make calls from my Grandstream phone to my X-Lite soft-phone,
the call does not complete correctly.
Scenario:
1. I take the GS off hook and dial 1100 (the extension of the
x-lite phone).
2. The x-lite
Aha. It may be connected to this error message, then:
messages:Oct 26 21:18:15 NOTICE[137382912]: File sched.c, Line 209
(sched_settime): Request to schedule in the past?!?!
I read somewhere that I could ignore this message, therefore I just
didn't include it in my earlier
report.
I just noticed that messages like this:
WARNING[1142106560]: File chan_sip.c, Line 451 (retrans_pkt): Maximum
retries exceeded on call
[EMAIL PROTECTED] for seqno 8338
(Response)
And this:
WARNING[1142106560]: File chan_sip.c, Line 451 (retrans_pkt): Maximum
retries exceeded on call
[EMAIL
I still don't understand why you can't just bind to
0.0.0.0 which means all interfeces. What problem does
that cause?
--- Asterisk [EMAIL PROTECTED] wrote:
The problem should be easy enough to solve for someone who knows the
internal guts. As a matter of fact, this is very important
Title: Message
Well , I've found something else..
It only fails with seg fault where the destination
h323 device is my cisco as5300..
when I'm using NetMeeting , it works
fine..
I'm gonna try it with others h323 devices , but not
right now..
Regards
Alex
- Original Message -
A lot of people put two NICs in a fire wall box for
security puroses. THat way your local network is
physically isolated from the public Internet. That way
no matter how dumb your users are and how badly they mis-
configure their Windows boxes you are protected. Without
a physically isolated
I'm not the original poster, but it certainly sounds like an underlying
OS config issue and not an asterisk issue. I'd bet a small amount of
money on multiple default gateways and/or associated routing issues.
I still don't understand why you can't just bind to
0.0.0.0
Title: Message
is
true i have the same problem with my vg200 and mc3810 ( cisco
device)
regards ,
Victor
Medrano
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexandru
CoseruSent: Tuesday, October 28, 2003 1:01 PMTo:
[EMAIL
I would like to try out RxFax as well. Where can I find it?
Thank you,
Christian Lademann.
On Tue, 28 Oct 2003 09:50:50 -0500
Johnson, Randy [EMAIL PROTECTED] wrote:
Here's one last plea for help from the list before I give up on the totally
cool software FAX modem concept in complete despair
Hi Brad,
Citeren Brad Waite [EMAIL PROTECTED]:
Why would you need another box? Couldn't you just provision one PRI card for
asterisk and the other for SS7?
Hmm, possibly. As I understand the SS7 stuff occurs at driver/module level, so
it might not be possible to link the zaptel driver to
Hi,
I just updated my image from CVS, compiled and reinstalled it. Now
whenever I make calls from my Grandstream phone to my X-Lite soft-phone,
the call does not complete correctly.
Scenario:
1. I take the GS off hook and dial 1100 (the extension of the
x-lite phone).
2. The x-lite
Hi,
I looked into include/asterisk/cdr.h, and tested a
few local transfers ( extension to a extension )
but I was unsuccesful in finding how a basic
call transfer would be logged. It looks as if
the only data available is the call origin and
first hop. No data as to who the call was transfered
What is better?
Cisco 7960 or Snom 200 ??
Bartosz
On Tue, 2003-10-28 at 14:28, Christian Lademann wrote:
I would like to try out RxFax as well. Where can I find it?
Try www.oncall.org.
software at ftp://ftp.opencall.org/pub/spandsp
Instructions at http://www.opencall.org/instructions.html
On Tue, 28 Oct 2003 09:50:50 -0500
Johnson, Randy
Are there any companies/consultants in the Indy area that
are Asterisk experts? Please contact me via email. THanks.
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http://lists.digium.com/mailman/listinfo/asterisk-users
Wouldn't it be interesting to combine this with the discussions on putting
voicemails in imap?
Use the mailbox for both, and you can store faxes and voicemails in the same
mailbox. Just make voicemail skip it with message 3 is a fax or something
similar.
-G
- Original Message -
From:
I prefer 7960.
Works out of the box with no problems, it
is sexy and after all it is Cisco gear.
Ta
Senad
Title: Message
both
works fine , cisco expensive .. snom 200 GOOD
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
JozwiakSent: Tuesday, October 28, 2003 3:40 PMTo:
ASTERISK USERSSubject: [Asterisk-Users] Cisco or Snom
Hi,
has somebody tried to get a Scitel-Brix QE ISDN card to work with
asterix ? I would like to know before I try...
Thanks
Patrik
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I'm having a problem with Asterisk picking up the Zap/1 and thinking its
a new call when instead I've already been on the phone talking to a
person. This is not my ideal setup and currently I have just an FXO card
and Asterisk is in parallel with my phone system instead of being in the
front.
I'm having a problem with Asterisk picking up the Zap/1 and thinking its
a new call when instead I've already been on the phone talking to a
person.
I had the same problem and used callprogress=no in the zapata.conf
file corrected the problem. It's a fairly well known problem where *
is
Hi everybody,
I have 3 TDM400P installed in a machine,and though the
4 ports of the first card work fine, some ports on the
other two have low or no signal and a noise instead.
Can someone help?
Thanx
__
Do you Yahoo!?
Exclusive Video Premiere - Britney Spears
Bartosz Jozwiak wrote:
What is better?
Cisco 7960 or Snom 200 ??
Bartosz
How much do you want to spend and do you really want the name??
Haven't used the Cisco, way too pricey..
Snom's work great..
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[EMAIL PROTECTED]
Jim,
What type of cabling are you using? What's terminated on the other end of
each port (Channel Bank, Telco Demarc?) How far away are you from what's
connected on cards 2 3?
This will have a lot to do with signal and noise?
-sb
-Original Message-
From: Jim Paraschou
It is a cable 4-5 meters long that has handssets
connected
I don't think its a matter of a distance
__
Do you Yahoo!?
Exclusive Video Premiere - Britney Spears
http://launch.yahoo.com/promos/britneyspears/
___
my mistake, I was thinking a T-1 card.
-sb
-Original Message-
From: Jim Paraschou [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 28, 2003 4:57 PM
To: [EMAIL PROTECTED]
Subject: # [Asterisk-Users] TDM 400P signal problem
It is a cable 4-5 meters long that has handssets
connected
I
Hello everyone and welcome to my first post to the list!
After studying for a couple of weeks, I finally built * for the first
time last night, and of course had the same SIP-behind-NAT woes that
plague all of us who use NATted connections.
It was therefore with no small joy that I read the fix
Hi list,
I'm trying to connect a cisco 2620 to my asterisk box using ISDN PRI.
But I got some problems.
zttool shows no alarms and Internally clocked. Asterisk starts
fine, but doesn't bring up d-channel. And when I try to make a call,
asterisk shows:
Honestly I can't see all these NAT woes people speak of... I have * on a
public ip .. sip.conf entries with nat=yes load em up.. and they work. So
I have yet to see why everyone has SO MANY problems.
bkw
On Tue, 28 Oct 2003, Christopher Stephens wrote:
Hello everyone and welcome to my first
I finally got this to work without crashing * but the resulting tiff file
is 8bytes
http://www.bkw.org/~brian/rxfax.txt
No fax... maybe that can help.
bkw
On Tue, 28 Oct 2003, Steven Critchfield wrote:
On Tue, 2003-10-28 at 14:28, Christian Lademann wrote:
I would like to try out RxFax as
What about the Grandstream Budgetone 101 at $65.00
How do these compare?
--- WipeOut [EMAIL PROTECTED] wrote:
Bartosz Jozwiak wrote:
What is better?
Cisco 7960 or Snom 200 ??
Bartosz
How much do you want to spend and do you really want the name??
Haven't used the Cisco, way
I had the same problem and used callprogress=no in the zapata.conf
file corrected the problem. It's a fairly well known problem where *
is sensing voice on the pstn line (from the analog phone), and initiates
the inbound call.
(If you talk quitely, it won't happen. Talk loud and the phone
What about the call quality between the cisco and the Snom on the
handset and speakerphone? Has anyone had experience using both and
noticed that the quality better or worse on one or the other.
i.e. - I'm not sure if the Snom is full-duplex speakerphone?
I'd suggest most speakerphones are
I have an X100p cardand it is hard to hear the person on the other
end. Should I mess with these values? I have heard both yes and no to
this question in the past. If yes, how much louder should I make them?
Thanks,
MIchael
___
Asterisk-Users
Obviously the prices are by no means, comparable. However, I'd go with the
Cisco 7960 - from my experience, they've been incredibly easy to configure
and the user interface is slick. Along with XML capabilitites, are office
'geeks' are enjoying them (that includes me.) Sound quality is great
Hi,
I am using the manager interface to make a web-based interface to making
calls.
Our system is configured to dial a certain number to get an outside line,
you then hear the dial tone of the PSTN and dial your number.
I would like to know if it is possible to do this via the manager
Hi,
I'm wondering if there's a way within a dialplan or AGI to find out
if an extension (SIP client) is already in use and the
person is already on the phone?
By default the channel is assumed available and callwaiting tone
is transmitted to the called extension. AFAIK there's no way to turn
Hi,
I am using the manager interface to make a web-based interface to making
calls.
Our system is configured to dial a certain number to get an outside line,
you then hear the dial tone of the PSTN and dial your number.
I would like to know if it is possible to do this via the manager
w only works on Zap channels, as far as I know.
On Tue, 2003-10-28 at 17:42, David Broker wrote:
Hi,
I am using the manager interface to make a web-based interface to making
calls.
Our system is configured to dial a certain number to get an outside line,
you then hear the dial tone of
Michael,
I've added a patch a week ago on to bugtracker to fix this - feel free to
try it and let me know
http://bugs.digium.com/bug_view_page.php?bug_id=408
Paul
- Original Message -
From: Michael Ulitskiy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, October 29, 2003
Hello,
I tryed out spandsp with libtiff-3.5.7 and with Asterisk from CVS.
I tryed to receive a fax on a CAPI channel. Finally I got a file with
8 byte length (/tmp/testfax.tif).
How can I do next?
Thanks in advance,
Thomas
ps: what are hardware requirements for sending/receiving
Ya dont say.. same problem here! :P
On Wed, 29 Oct 2003, Thomas wrote:
Hello,
I tryed out spandsp with libtiff-3.5.7 and with Asterisk from CVS.
I tryed to receive a fax on a CAPI channel. Finally I got a file with
8 byte length (/tmp/testfax.tif).
How can I do next?
Thanks in
Hello,
The annoying one for two questions:
1) when I recover messages, * jumps of extension.conf toward Voicemail.conf
and access to some options default of *.
The question is:
Can I modify those options?
Where ?
How?
2) If I want to make a change of password in
On 27/10/03 21:57, DUSTIN WILDES wrote:
Does anyone have any recommendations on implementing Answering
Machine detection for call generation programs?
There's obviously no nice way of doing this.
If you're doing telemarketing, and you're playing pre-recorded audio,
which of course is a nasty
On 28/10/03 08:17, Florian Overkamp wrote:
look for a T1 failover switch.
Nice tool. Anyone know of an E1 equivalent ? :-))
Most people who'd want this sort of thing probably have multiple
incoming E1 lines. If you have multiple lines, you can set up a hunt
group to range over the lines, with
Hi there,
can someone point me to a resource where I can find a list and/or
description of * built-in functions like transfer, call waiting etc?
I know I can transfer with #, and read about *8 for call pick-up, but how
do those (and more) exactly work, and what else is available I haven't
Does anyone know of a command-line tool that I can use to mix my own MOH
tracks? Specifically, I want to be able to do this:
1) Record a Your call is valuable to us... advertisement
2) Specify a number of song files to be played randomly/in sequence/whatever
3) Insert or overlay the
Grandstreams phones can't call out with the latest CVS, anyone know what the
last good CVS date was?
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Howdy y'all,
I am using an ATA-186, through Vonage, connected to my X100P. It works
well - except, hanging up... The hangup doesn't register, most of the time,
with Asterisk. Are there any known tweaks out there that I should look
into? I've noted some have altered dsp.c, and rebuilt. Just
Humans tend to say Hello? (short burst of audio followed by silence),
and answering machines tend to say I'm sorry I'm not here right now,
please leave a message after the beep (long burst of audio followed by
a beep and silence).
So, basically you need to decide 1) what is audio and what is
Does anyone know where I can buy SNOM or Cisco (new
or used) phones the cheapest. I need a few
Todd Wallace
WipeOut wrote:
Bartosz Jozwiak wrote:
What is better?
Cisco 7960 or Snom 200 ??
Bartosz
How much do you want to spend and do you really want the name??
Haven't used the Cisco, way too pricey..
Snom's work great..
You have to look at what CODEC support you want also.
My Cisco 7960 phones
Todd Wallace wrote:
Does anyone know where I can buy SNOM or Cisco (new or used) phones
the cheapest. I need a few
Todd Wallace
Uh http://www.ebay.com/
-Andrew
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[EMAIL PROTECTED]
I have been watching for SNOM phones on ebay and have not seen any. There
are plenty of Cisco phones, so I can definitely price those there.
Todd Wallace
- Original Message -
From: Andrew Gillham [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 28, 2003 8:39 PM
Its not an issue with CVS my grandstream works fine.. what kind of errors
are you getting?
bkw
On Tue, 28 Oct 2003, James Sizemore wrote:
Grandstreams phones can't call out with the latest CVS, anyone know what the
last good CVS date was?
___
Everyone,
Just thought I would drop a line telling everyone here I have the software
RxFAX/TxFAX up and running without any real problems. I did have to.
RH 9.0
1) Install an audio devel rpm
1) install libtiff from source, and copy over a bunch of include files to
/usr/local/include
2)
Good for you... All I can get are 8 byte tiff files.
On Tue, 28 Oct 2003, Brian Schrock wrote:
Everyone,
Just thought I would drop a line telling everyone here I have the software
RxFAX/TxFAX up and running without any real problems. I did have to.
RH 9.0
1) Install an audio devel
Does anyone know of a command-line tool that I can use to mix my own
MOH tracks? Specifically, I want to be able to do this:
1) Record a Your call is valuable to us... advertisement
2) Specify a number of song files to be played randomly/in sequence/whatever
3) Insert or overlay the
Grandstreams phones can't call out with the latest CVS, anyone know what the
last good CVS date was?
You may be experiencing difficulty due to bad codec permissions,
since the latest CVS updated version (earlier today) does seem to
work for me. Make sure you have correct allow/disallow
This would be why it works for me.. I specified the codec for the phones
on a per peer basis.
On Tue, 28 Oct 2003, John Todd wrote:
Grandstreams phones can't call out with the latest CVS, anyone know what the
last good CVS date was?
You may be experiencing difficulty due to bad codec
Just as a heads up, soon, I will be merging Thorston Lockhart's new tagged
CVS archive over to Digium. This will mean you have to do a *clean* check
out of asterisk, zaptel, libpri, etc.
For those of you with *localized changes*, please be sure to do:
# cvs diff -u ../my-asterisk-changes.diff
As for *why* we're doing this change, and what advantages there
will be and how to use them, Thornston will send that information in a
separate e-mail.
Back in February, the original CVS repository was lost. With that, all
change history
from before then was lost as well.
I have
That's for pointing out Walter Snel hack.
Adding his two additional features would not be
hard.http://lists.digium.com/pipermail/asterisk-users/2003-October/024968.html
in sip.conf nat=1 means the _client_ that Asterisk is
talking with is NAT'd. We could add a line like below
to sip.conf
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