Lal, Deepak (Contractor) wrote:
When I get a SIP call, I get the following error:
*CLI NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is
'multipart/mixed;boundary=unique-boundary-1', not 'application/sdp'
Which client is used to place the call? I haven't seen
Shoval Tom wrote:
This problem exists with all of the DNS servers I tried.
I tried several ISPs in Israel and a couple at the US.
; DiG 9.2.1 www.voip-info.org
;; ANSWER SECTION:
www.voip-info.org. 3600IN A 192.168.168.3
This is mystical. If I dig in 192.116.202.99, the
--unique-boundary-1
Content-Type: application/ISUP;version=cp10isup;base=etsi121
Content-Disposition: signal;handling=optional
01 07 02 70 00 02 01 03 09 02 0a 00 0a 07 03 13 15 44 12 01 20 04 08 83 10 15 74
77 11 11 0f 06 01 10 00
--unique-boundary-1
Hi!
Content-type: application/ISUP
---
mtm spm wrote:
Hello,
I have a problem with asterisk when dial out to a SIP
provider.
Asterisk send a INVITE with no credentials, the
provider reply with a 401 Unauthorized.
However, Asterisk DOES NOT resend the invite again
with credentials. But it hangs there (maybe waiting
for a ok)
It is
John Todd wrote:
exten = 514777,1,Dial,Zap/2|10
Try:
exten = 514777,1,Dial(Zap/2,10)
I think these two versions of giving arguments are confusing. Reading docs
and show application texts, both variants are used, sometimes even in the
same text.
Is the first syntax old, to be
Interesting. Can you point to where this is documented? I rooted around
thru the Digium online manual, whitepaper, etc, couldn't find any doc.
http://www.voip-info.org/wiki-Asterisk+variables
/O
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Louis-David Mitterrand wrote:
On Thu, Nov 06, 2003 at 10:41:15PM -0500, Brian Schrock wrote:
Hello,
I have searched google, read everything on the mailing list, read
/usr/src/asterisk/README.iax and /usr/src/asterisk/doc/iax.txt(?), asked on
the IRC channel and I cannot figure out what is wrong
Michael Manousos wrote:
when i try to make a call with netmeeting through * ( * dial out with
Dial,OH323/[EMAIL PROTECTED] ) the call will be blocked.
This is a problem of OpenH323 1.12.0. Use this dial string:
Dial,OH323/h323:[EMAIL PROTECTED]
Or, even better, use the latest (it has been fixed).
C M wrote:
hi,
i saw the cdr file called Master.csv and i want to
know what these represent. examples
,,4,incoming,,Zap/1-1,Zap/4-1,Voicemail,u,2003-11-07
17:43:04,2003-11-07 17:43:04,2003-11-07
17:43:22,ANSWERED,DOCUMENTATION
,,19373693874,incoming,,Zap/1-1,IAX[Voicepulse]/1,Dial,IAX2/[EMAIL
mtm spm wrote:
Hi Olle,
--- Olle E. Johansson [EMAIL PROTECTED] wrote:
The first Invite is without credentials, since
digest authentication needs input
from the server to create credentials.
This is also what I understood too from rfc.
I was just confused becouse in the Asterisk code
mtm spm wrote:
Maybe this is a silly question but I am a beginer with
Asterisk.
I want now to be able to write a script or something
so that I can dial out a number and when the call is
answered to play a .gsm file or an output from
festival.
I need to call this numbers on demand(from another
Mireia Munoz de jesus wrote:
Hi!
I am trying to know well asterisk. For that I would like to know the exact role
for each config file. Can someone tell me what is the role of the next ones or
a web where I could find this information? That will be very helpful.
- alsa.conf
- enum.conf
-
Steven Critchfield wrote:
We have to rename Zaptel timing to Asterisk timer, which is more correct
since there are several ways of getting a timer to work, only one of them
is by using Zaptel cards.
http://www.voip-info.org/tiki-index.php?page=Asterisk+timer
Actually it needs to be zapata
Steven Critchfield wrote:
On Fri, 2003-11-07 at 16:04, Olle E. Johansson wrote:
Steven Critchfield wrote:
We have to rename Zaptel timing to Asterisk timer, which is more correct
since there are several ways of getting a timer to work, only one of them
is by using Zaptel cards.
http://www.voip
I've stopped referring people
directly to my Asterisk site and instead refer them to the Unofficial
Links page at Digium.
--Eric
--
Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/
;-)
/O
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The great thing with Open Source is that anyone is free to go away and
create a Red Hat-ized version of the software. As long as the GPL is
followed by the book, everyone should be happy. Don't just expect everyone
to join you.
There's a lot of people on the mailing list who in a small form have
a
Mark Spencer wrote:
The OpenOffice.org project have a marketing subproject that has been
doing a lot of non-development things. Coordinating pressreleases,
arranging conferences, participation in other conferences, training
material etc. Maybe we should look into stealing ideas from that project?
hkirrc.patrick wrote:
i've just spent the pass 2 days trying to get AGI to work with PHP;
i made a lot of silly mistakes along the way which could have been
avoided if only there were some kinda howto or samples. at the risk
of looking stupid, i decided to shared my experience in hopes that
it
Florian Overkamp wrote:
Hi,
At 13:48 11-11-2003 +0800, you wrote:
i made a lot of silly mistakes along the way which could have been
I've done this a long time ago, and by sheer accidence someone posed a
question about it yesterday. Below are some of the comments I sent him.
...and I've
Alexandru Coseru wrote:
Hello..
Maybe I'm asking something silly but. How can I register my own app
with * ?
Look in modules.conf
http://www.voip-info.org/wiki-Asterisk+config+modules.conf
(New page, just found out that it wasn't there...)
/Olle
I've tried to summarize my experiences of Asterisk on a FreeBSD system:
http://www.voip-info.org/tiki-index.php?page=Asterisk+FreeBSD
If you have facts to add, please mail me off list or edit the Wiki page.
/Olle
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Freddi Hansen wrote:
with boards from Aculab, we are replacing Aculab boards with Digium
boards BUT we would need more
Digium boards IF we could use both Digium and Aculab cards in the same
server. The reason being that
TE410P doesn't support SS7-ISUP so we continue using only Aculab cards
in
On request, I've updated the following page
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20indications.conf
with ring signals from Brazil. (And at the same time, the Brazilian signals was added
to the CVS).
If you have an entry in indications.conf that you want to share, a
Andrew Joakimsen wrote:
I am trying to get Asterisk to compile on FreeBSD 4.8. Per bug 389, BSD
support should be in CVS. I have also tried applying the patch in bug
374, but always get these messages:
click# make
Makefile, line 21: Missing dependency operator
Makefile, line 23: Need an operator
This is such a wonderfully clear explanation of the asterisk sequencing
methodology that I would suggest including it in the documentation as an
example for using include in contexts.
Stephen Besch
Agreed.
Added to
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf
[EMAIL PROTECTED] wrote:
My problem is that when I dial into extension 1000 it is telling me this
is not a valid conference number. Can anybody telling me what I'm doing
wrong here?
I've added the solution to this to the FAQ:
http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ
/Olle
Philipp von Klitzing wrote:
Hi!
I'm confused as to where I need to place the Asterisk server with repect
to my Linux Firewall? I've read thru the message archives but have not
been able to glean a clear answer. My Linux Firewall is a RH9 running
IPTABLES doing NAT.
The probably easiest
With the assistance of writers Stephen and Rich, I've edited two new pages on the Wiki.
It's general information on Echo cancellation - how to attack it and locate it.
You'll find them from the FAQ:
http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ
Still looking for more advice, tips and
WipeOut wrote:
Philipp von Klitzing wrote:
Hi there,
I still have issues with the IAX connection between two servers (one
static (server A), one dynamic (server B), none behind NAT):
B registers with A, and iax2 show registry shows that everything is
fine. However, after a while if I check
Areski wrote:
Hello All !!!
I trying to make meetme working!
I don't have zaptel interface and I cannot install ztdummy cause I don't
have usb-uhci !!!
Is there any way to get it working ???
FAQ
http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ
See the page on Zaptel timers, called
David Carr wrote:
We have a cron job that moves master.csv to subdir/timestamp.csv and then
tries to parse and delete all csvs in that directory. The reasons we did
this were
1) We wanted to store more data in the database without having to change
source code. For example, we have 45 asterisk
SIP Express Router have radius support. Look there for hints on how to get
Radius support for VOIP.
http://iptel.org/ser/
/O
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Added to the top of the FAQ.
http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ
/O
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Florian Overkamp wrote:
Citeren Daniel ANDRE [EMAIL PROTECTED]:
Do you have any information on how to include these functions to th
dialplan?
Depends on what you'd want precisely. Here's a (higly undocumented) example
(I'm working on more). I used KPJ's examples on www.junghanns.net to start
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+CallingPres
Could someone explain this applicatoin a bit more? I found the application in the Zap
channel source,
and a comment says something about PRI connections. What is the value specifying?
Thank you!
/Olle
Mark Spencer wrote:
Why don't we just add it on the DIgium list server, wouldn't that make
more sense, to have a single place for all list memberships?
Yes, please. Doing that makes it easier to find it.
/Olle
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Robert G. Werner wrote:
Btw, I encourage those of the Wiki readership who can spell their way
out of a wet paper sack to not hesitate in fixing typos. Mis-spellings
really do make docs somewhat suspect, to some types of people. Not me,
of course. ;-)
Thank you! My native language is not
Even more cool is to start using ENUM. There's a good new article on how to start doing
that on the Wiki, not contributed by me.
Since the ENUM tree is not very active, only experiments in some countries, we could
start
building our own Asterisk/IAXtel ENUM-like tree. One problem though is that
The amount of mail on asterisk-users is more than even *I* can read in a
day, and my job is 100% asterisk. There probably is a justification for a
new list, but I think it is less the -biz list as much as much as the
-newbies. Keeping a business discussion on -users is probably quite
useful
Internationally, there is already an officially sanctioned country code
for Universal Telecommunications Services, and it's +878. There is
quite a bit of activity now in moving that area code from the ITU
sanctioning (which happened a few weeks ago) and now moving towards
commercial
I will say that the Wiki is very hard to deal
with as getting information out of it! It tends to go in the wrong
direction allot! Some of us just don't have the time to go through it!
Could you please elaborate a bit more, to help us steer the wiki in the
right direction?
As I see it, the Wiki
But that would sort of break SIP. A SIP URI is [EMAIL PROTECTED], so it makes
No, A SIP URI is [EMAIL PROTECTED] - there's a big difference. Read on
DNS SRV records on
http://www.voip-info.org/tiki-index.php?page=DNS%20SRV
Quoting myself:
No one really mails [EMAIL PROTECTED] any more. We're
Olle, are you watching, this is for the Wiki.
I'm here, trying to catch up :-)
Don't forget that applications are also modules and can be set to not
load. I don't list applications here as they have been listed elsewhere.
http://www.voip-info.org/tiki-index.php?page=Asterisk+modules
Thank you!
John Todd wrote:
At 11:47 AM -0600 11/20/03, Tilghman Lesher wrote:
(SIP, Zap, whatever) has their own CLASS dialplan sets, then that
is a different problem - either deactivate them and use the server,
or leave them enabled and ignore things for that line.
How exactly did you hack your
That said, I find an FAQ quite a good idea. Maybe just as another page on
the voip-info.org Wiki?
http://www.voip-info.org/wiki-Asterisk+FAQ
It's been there for a while now.
Thank you, anyhow, for suggesting improvements.
/O ;-)
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Rich Adamsson and I have started a new Wiki page to document configuration for
different VoIP clients - both hardware and software.
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phones
Rich started with writing documentation on the Cisco 79xx phones.
Please help us adding information
I just discovered that the SIP channel has undergone some major improvements.
I'm now able to dial any SIP URL with dial, couldn't get it to work earlier,
all domains had to be defined in SIP.conf.
This, in addition to the SIPDOMAIN variable, makes the SIP channel even more
useful.
Thank you,
Olle E. Johansson wrote:
I just discovered that the SIP channel has undergone some major
improvements.
I'm now able to dial any SIP URL with dial, couldn't get it to work
earlier,
all domains had to be defined in SIP.conf.
...and I'm able to call any SIP URL with Xlite, with Asterisk resolving
C M wrote:
the real
problem is with the asterisk NAT issue. i was asking
for help if any one had similar problem with nufone
account. i am using IAX. is there anything like
nat=yes as in sip.conf?? i read iax should work with
normal configuration. its ok with outbound. i only
have problems with
If you update your source from the CVS, you'll get a new SIP channel
that supports a new syntax for SIP calls in extensions.conf
If you define a SIP peer in sip conf, like
[mysipprovider]
...
You can now use
dial(SIP/mysipprovider/extension)
Where the part mysipprovider is related to the
[EMAIL PROTECTED] wrote:
Hello,
I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323.
I would like to add this to the Wiki, but wonder which product you mean in
Cisco's product range?
/O
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Steve Underwood wrote:
Steven Critchfield wrote:
On Tue, 2003-11-25 at 09:24, Jerimiah Cole wrote:
Does anybody have useful tips on creating good quality recordings for
use with prompts in asterisk? I'm interested in hearing input on
hardware (mics, dats, sound cards, etc) and software
Asterisk wrote:
Hello!
Does anyone know where I can find out about the CDR fields?
I know most of them are self expiatory, but what is disposition for?
I've done a search in Google, I even went to dictionary.com to check the
meaning of the word, but I don't know why it always equals 4 in my
Angel Gabriel wrote:
Does anyone know if a web interface has been created for * ?
There's one in the source tree, phpconfig. There's a lot of other projects
out there, depending if you want
* A configuration interface
* A user interface
* A receptionist/manager interface
See
Anyone that knows if the Asterisk SIP channel supports symmetric RTP?
/O
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Just found this IETF draft, that gives a lot of examples on how to implement
PBX features in SIP. Good inspiration!
http://www.voip-info.org/tiki-index.php?page=SIP+PBX+functions
/Olle
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Walker Haddock wrote:
On Wed, Nov 26, 2003 at 08:33:13PM +0100, Olle E. Johansson wrote:
Asterisk wrote:
Hello!
Does anyone know where I can find out about the CDR fields?
I know most of them are self expiatory, but what is disposition for?
I've done a search in Google, I even went
Jan Janak wrote:
On 26-11 22:16, Olle E. Johansson wrote:
Anyone that knows if the Asterisk SIP channel supports symmetric RTP?
Yes.
Followup question:
Both as a SIP UA (Client) and as a SIP proxy?
/O
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Jan Janak wrote:
On 27-11 15:14, Olle E. Johansson wrote:
Jan Janak wrote:
On 26-11 22:16, Olle E. Johansson wrote:
Anyone that knows if the Asterisk SIP channel supports symmetric RTP?
Yes.
Followup question:
Both as a SIP UA (Client) and as a SIP proxy?
I don't know, I tried asterisk
Anyone succeeded in using regexp replacements in ENUM, like
!\\+421257296(.*)$!sip:[EMAIL PROTECTED]
I can't get it to work in ASterisk.
I've added '\\1' and Debug echos 1
I've added '1' and debug echoes \1, but regexp fails to work.
The example above is from the nic.at presentation, I can't
Brian West wrote:
On Thu, 27 Nov 2003, Olle E. Johansson wrote:
Olle E. Johansson wrote:
Anyone succeeded in using regexp replacements in ENUM, like
!\\+421257296(.*)$!sip:[EMAIL PROTECTED]
I can't get it to work in ASterisk.
I've added '\\1' and Debug echos 1
I've added '1' and debug
Hcqm wrote:
CAN I USE/COMPILE ASTERISK
without any telephone/sound card?
I only want to use it as a IP PBX.
Yes, please go ahead.
I'm running Asterisk on both LInux and FreeBSD servers
without any PSTN or ISDN hardware.
Have fun!
/Olle
___
Andrew Kohlsmith wrote:
CAN I USE/COMPILE ASTERISK
without any telephone/sound card?
I only want to use it as a IP PBX.
This is another topic covered quite often. Do we have this in a FAQ/Wiki
entry yet?
Thank you for the reminder, now it is:
Leif Madsen wrote:
On Thu, 2003-11-27 at 12:03, Mark Spencer wrote:
Yea, cause I used both Kphone and Windows messenger, and they
successfully registered (and subscribed i think) towards asterisk. Using
Kphone I even get a online status on all other users on the asterisk but
no interaction with
Hcqm wrote:
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Hcqm wrote:
CAN I USE/COMPILE ASTERISK
without any telephone/sound card?
I only want to use it as a IP PBX.
Yes, please go ahead.
I'm running Asterisk on both LInux and FreeBSD servers
ranga wrote:
I have one linux box running asterisk ( say 192.168.68.15 ) and second box
running partysip (say 192.168.68.6).
Now this is what I wanted to achieve.
The other sip server ( here partysip) may have many users registered. It
is not possible to make every user's entry into
ranga wrote:
I agree with you. But the issue is, how could I fix the domain name
variable? This should not be static. The target domain changes as per the
choice of the user that is connected through softphone. For example, you are
connected to edvina.net. Now I want to call you from my
Roy Sigurd Karlsbakk wrote:
how about IAX2 trunking? does this work with ztdummy?
I was using both IAX2 trunking and MOH before getting my zap devices,
and I never had any luck with ztdummy.
Are you sure you had trunking up without it? I'm positive I've been told
trunking requires a timing
ranga wrote:
Olle,
These are the two lines that I added to extensions.conf.
exten = evaro,1,Dial(SIP/[EMAIL PROTECTED])
exten = john,1,Dial(SIP/[EMAIL PROTECTED])
I called sip:[EMAIL PROTECTED] first and then sip:[EMAIL PROTECTED] Here
is the console catpure.
-- Executing
In my expeditions into ENUM I've discovered that using old BIND servers leads
to problems with the \\1 construct. Zone transfers between old BIND servers
that works for all other DNS functions leads to \\1 being converted to
\1 and your regexp fails. The zone on the primary and secondary DNS
Iain Stevenson wrote:
I've been tinkering with ENUM and found that the lack of a debug message
in enum.c that says it has actually succeeded in resolving an address is
a bit of a nuisance. It makes it difficult to see if failures with ENUM
are due to problems with parsing NAPTR records (in
Joe Dennick wrote:
I think you need to better define your Queue Environment in
extensions.conf. Below is what I've got in mine, and it seems to work
quite well:
exten = s,1,Answer
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
--
*** Olle E. Johansson, [EMAIL PROTECTED]
Mobile +46 70 593 68 51, Edvina AB, http://www.edvina.net
Runbovägen 10, 192 48 Sollentuna, Sweden
Phone: +46 8 594 78 810, Fax: +46 8 594 78 820
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http
ranga wrote:
This is the complete extensions.conf. I wasnt getting the SIPDOMAIN right.
Rest of your script/configuration works only if ${SIPDOMAIN} works
Am I missing anything in this? I had the latest CVS checkout this morning,
i.e., 1st Dec. 12.00 Noon GMT +5.30.
Ranga,
I agree, seems like the
Cees de Groot wrote:
zoa [EMAIL PROTECTED] said:
And while you are in Europe, why not also do Brussels ? ;)
Amsterdam!!
Feel forced to add STOCKHOLM!
/O ;-)
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[EMAIL PROTECTED]
Vutukurusip:[EMAIL PROTECTED];tag=21632105
To: sip:[EMAIL PROTECTED];tag=as62db81f5
Via: SIP/2.0/UDP 192.168.68.12:5060
7 headers, 0 lines
localhost*CLI
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 01, 2003 2:16 PM
Subject: Re: [Asterisk
localhost*CLI
- Original Message -
From: Olle E. Johansson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 01, 2003 2:16 PM
Subject: Re: [Asterisk-Users] Asterisk as SIP Proxy
ranga wrote:
This is the complete extensions.conf. I wasnt getting the SIPDOMAIN
right.
Rest
Bartosz Jozwiak wrote:
I would like to release prepaid application.
But I have a small problem, we are using their Cisco prompts (nice lady
voice)
And I do not know if it is ok to release it.
I don't know about the rights to the Cisco prompts, so be sure
to remove them and then release it. User
Alastair Maw wrote:
On 28/11/03 07:39, Olle E. Johansson wrote:
The latest version of my Ethereal plugin for IAX2 is now available here:
- http://almaw.com/ethereal-iax2-plugin-0.3.zip
Could you please create a URL that is a bit more non-version-specific?
http://almaw.com/etheral-iax2/
It now
Steven Critchfield wrote:
On Tue, 2003-12-02 at 07:44, Philipp von Klitzing wrote:
BTW: Where exactly is the difference between Hangup and Softhangup()?
Hangup is something done in the course of the dialplan and works on the
current channel where softhangup is a cli command that works on a
Brian West wrote:
If you have echo on the X100P's Mark setup chan_zap to pretrain the echo
can, but it had a few issues until today which Mark nailed down the bug
that caused the DTMF to be unreliable.
Ok here is how you would do it:
Thank you!
John Harragin wrote:
OK, an answer is in README.variables causes.h...
[7-digit-PRI-Machine-2] ; The machine connected to PRI 2 (on its g1)
exten = _9NXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _9NXX,2,gotoif,$[${HANGUPCAUSE} = 2]?9|1
exten = 9,1,Busy
Added to
jerk face wrote:
I recently switched from Mandrake to Redhat and I
noticed that vmail.cgi does not work with the default
apache installation that comes with Redhat.
Here is what I get in my error logs:
[Thu Dec 04 11:59:57 2003] [notice] suEXEC mechanism
enabled (wrapper: /usr/sbin/suexec)
[Thu
Please try my Asterisk search engine at:
http://search.voip-forum.com
for searching the list.
Any comments to me off list.
I'm indexing lists.digium.com voip-info.org and iptel.org
/Olle
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[EMAIL PROTECTED]
-Original Message-
From: TC [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 11:58 PM
I have had several instances over the last month of Asterisk freezing,
Does anyone have any suggestions? or ideas as to what may be causing it?
Sounds like some type deadlock
Take a look here
marrandy wrote:
On Friday 05 December 2003 09:47 pm, marrandy wrote:
In the meantime, I've pulled information that may, or may not be correct.
If
people can verify or add to this, it would be appreciated.
-
Codec
Rich Adamson wrote:
On Sat, 2003-12-06 at 07:45, rnc Info Lists wrote:
Is anyone other than me having trouble dialing out via IAXTEL? I havn't
changed my config files in weeks but seems that IAXTel calls (800 and FWD)
stopped working in the past week sometime.
To add to this problem, we
Wim Venneman wrote:
Has anyone have an idea why, if you capture the files on a Asterisk
network (ex with Ethereal) you always see the communication between the
two sip phones( hard or soft) passing through the asterisk server (on
UDP layer)
Yes.
Isn't SIP a protocol that (after that it has
Brancaleoni Matteo wrote:
SIP control messages goes always through the server
(port 5060) , only RTP media streams is p2p .
you can see RTP passing not p2p but by * server if:
* the phone doesn't supports reinvites
or
* set in sip.conf canreinvite=no in the user definition
or if the both ends
Yes, try using this instead:
Dial,IAX2/[EMAIL PROTECTED]/extension
Hmm, thats a thought, but I thought peername would need to be a host ?
Also, the nasty part is that specific portion from extensions.conf on the
master server is generated, so I'd need to change the code for it.
Usually
James Schenck wrote:
Even if I get no help from asterisk users I will get this to work I will
just post how I solved the problems for the benefit of others who seek help
and search the archives
If you don't want to help new guys or you think I should just figure it out
on my own you should just
http://www.voip-info.org/tiki-index.php?page=Asterisk+multi-language
By trial and error and a lot of ancient nordic magic (reading the source) I found out
that Asterisk does not look for language-specific sound files with the -cc extension,
cc being country code.
Asterisk looks for files first in
Nicolas Gudino wrote:
I'm not a GPL expert, so I have a few questions: Does an AGI script needs to
be distributed in source form? Maybe this application/script is using
Asterisk unmodified. They can sell just their AGI scripts and provide only
asterisk with full source?
An AGI script does not
Nicolas Bougues wrote:
On Tue, Dec 09, 2003 at 08:02:18AM -0600, Mark Spencer wrote:
I suppose trunk groups on SIP would be interesting.
As I understand, trunking in IAX is meant to transport voice packets
from several calls between two hosts in the same lower layer packet.
Are there
Alastair Maw wrote:
On 10/12/03 07:41, Chris Albertson wrote:
I'd prefer to run a local sendmail. Ths means you have a local
queue and the mail gets handed off quikly even if your
other server is down or slow.
A better solution would be an SMTP fowarding agent, such as ssmtp. I'd
prefer
Thinking about this problem I would like to point out the root cause of all this:
*** The Asterisk open source PBX is a success story ***
We are a growing crowd. New users keep joining the list all the time, experimenting,
installing, getting along.
Some of them are used to Open
Seems very strange. Check which asterisk you start.
Stop the asterisk and start the binary in the source directory with ./asterisk and the
connect
to that version and check CVS date.
If it's different, then you have several versions in your path.
My 2 cents...
/O
/mailman/listinfo/asterisk-users
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*** Olle E. Johansson, [EMAIL PROTECTED]
Mobile +46 70 593 68 51, Edvina AB, http://www.edvina.net
Runbovägen 10, 192 48 Sollentuna, Sweden
Phone: +46 8 594 78 810, Fax: +46 8 594 78 820
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Asterisk-Users mailing list
Timothy Costello wrote:
and somewhere (maybe on the wiki) should be a link to ESR's How to Ask
Smart Questions: http://www.catb.org/~esr/faqs/smart-questions.html
I know it's been posted to the list several times. It should be part of
the FAQ to read it before asking questions...
Added link
Tilghman Lesher wrote:
On Friday 12 December 2003 07:25, Dan wrote:
Hi,
It would be great if the IAX protocol will be able to tranfer fax
data (even converted in another format) between Asterisk boxes,
using low bandwidth codecs like GSM.
I know that this is possible only with the G.711 now
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