And when it does work, the console says:
Mar 5 02:07:08 NOTICE[9962]: chan_iax2.c:7065 iax2_poke_noanswer: Peer
'akralliax' is now UNREACHABLE! Time: 5
Mar 5 02:07:18 NOTICE[9962]: chan_iax2.c:6420 socket_read: Peer 'akralliax'
is now REACHABLE! Time: 3
The iaxcomm phone is on the same LAN, so
I didn't know how else to caption this.
I'm trying to play around with codec pass-through. I have two SIP
phones, both with g729, behind two Asterisk servers.
I set all the configs, SIP and IAX, to disallow=all; allow=g729 on
both servers.
But the originating server won't even try to call the
Can anybody explain how to use in ASTCC Userconfig, Sip Friends, IAX
friends and
what it does, when you setup multiple trunks in routes?
bye
Ronald
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Dear all
I am get the following problem when trying to compile app_meetme2 using
mysql...it seems to want to use pgsql.? anyone
my Makefile looks like
app_meetme2.o: app_meetme2.c
#$(CC) -pipe $(CFLAGS) -c -o app_meetme2.o app_meetme2.c
$(CC) -pipe -I/usr/local/include/mysql
The error messages are Postgres related.
You need to have a special postgres include file (postgres-dev files) to make
it compile or disable postpres support somehow.
I'm using debian and the the concering include file resided in a subdirectory
of what asterisk was told.
Jens
Quoting Jay Milk [EMAIL PROTECTED]:
In a word - No. Generally, BT-capable phones can only control a headset
or handsfree-set, but not be turned into a headset themselves. It's
akin to expecting to watch TV on your remote, as it controls the TV so
nicely :)
Thanks, that's exactly what I
Am Samstag 05 März 2005 07:58 schrieb Remco Barende:
Hi list!
While looking for the Snom 190 I found another phone, the Elmeg IP 290
(www.elmeg.de).
Looking at the pictures the specs they seem to be very similar beasts
but the firmware is supposedly not interchangeable.
Does anyone know
Quoting dean collins [EMAIL PROTECTED]:
I believe the advice that was given to you 5 minutes after you posted
your original question was good and valuable and you should utilize
bluetooth in this format, otherwise - go read the wiki.
The Wiki only covers using the phone as an FXO via
At 04:34 AM 3/5/2005, you wrote:
The error messages are Postgres related.
You need to have a special postgres include file (postgres-dev files) to make
it compile or disable postpres support somehow.
I'm using debian and the the concering include file resided in a subdirectory
of what asterisk was
I have used the Draytek 2600V router in a few locations where only 1 or 2
phones are required.
The router has 2 FXS ports and can be used locally to an * box or via the
VPN to a remote * box.
The VPN built into the routers just works, and I have 1 user who has had 3
VPN circuits up and running
Hi,
I have a problem using AGI cmd SAY DIGITS. For some reason I cannot
here any thing when the script got executed. However if I use the cmd
SAY NUMBER I can here * reading the number fine.
I am running asterisk-1.0.6 and below is my PHP script.
Help please.
- Natt
#!/usr/bin/php -q
?php
Date: Fri, 4 Mar 2005 17:37:16 -0500
From: John Scully [EMAIL PROTECTED]
Subject: [Asterisk-Users] Is anyone using asterisk in a small call
center
Hello - I have just joined the lists and am considering installing quite a
few * systems.
I am looking for an IP-PBX with both solid standard
Asterisk 1.0.3
Sayson 480i running .78 release
(problem may not be Sayson specific, it's just that's what's deployed)
Problem: Asterisk rejects registrations every so often even though
nothing has changed either with Sayson or Asterisk configuration (and
previous registrations have succeeded)
On 5 Mar 2005, at 06:58, Remco Barende wrote:
Hi list!
While looking for the Snom 190 I found another phone, the Elmeg IP 290
(www.elmeg.de).
Looking at the pictures the specs they seem to be very similar
beasts but the firmware is supposedly not interchangeable.
Does anyone know the
hi!
i got the following problem:
the * is registered to a provider - sip.conf ... register =
user:[EMAIL PROTECTED]
sometimes it happens that the server goes down. when i try to make a
call it is not working ... he tries to reregister at the SAME server,
times out and tries to
Hi Again,
I used my phpconfig setup for a week, and found it a great timesaver for me
:-)
However I have just gone and broken it, and can't seem to 'fix' it.
I was running a xorcom rapid installation, but converted to a semi-standard
debian by changing the apt sources; so I could install a
On March 5, 2005 01:39 am, Jonathan Hobbs wrote:
Ignore them and they will go away.
Only after polluting the list with incessant How do I do X? messages, and
then only after subsequently polluting the list with asterisk sucks
messages, and then all the bad karma of some clueless twitt who
However now I cannot even browse a .conf file via phpconfig. When clicking
on the file I get the following error:
Warning: fopen(/etc/asterisk/iax.conf): failed to open stream: Permission
denied in /var/www/phpconfig/cls_phpconfig.php on line 127
I have gone over the wiki page, done chmod
Thanks,
I had thought this, and done the command:
chmod -R a+w /etc/asterisk
And it still didn't work.
However I just set chmod 777 via WinSCP recursively, and it worked :)
This is only a testing box I am not worried about the security risks.
Strange the chmod didn't work I feel?
C
I've not released the source yet, I asked last week on the mailing
list for people to send me over some example queue_logs, because so
far I've only been able to test the software against my own.
I have however made a lot of changes to it since last I posted about
it.
How is the progress on
Hi,
I want to
automatically send the sound files generated by asterisks monitor functions to a
certain email address. My knowledge of shell scripting leaves a lot to desire,
so I was hoping maybe on of you guys already did this and might provide me with
an example of what to do :)
Best
Hi All,
Googling X100P Clone I found several information
about these cards and seems that some winmodem has the same chip used from the
original X100P.
Here below a list winmodem which should work as
X100P clone:
1057 Motorola5608 SM56 PCI Fax Voice
ModemE159 Tiger Jet Network Inc0001
Hello Remco,
On Sat, 5 Mar 2005, Remco Barende wrote:
Hi list!
While looking for the Snom 190 I found another phone, the Elmeg IP 290
(www.elmeg.de).
Looking at the pictures the specs they seem to be very similar beasts
but the firmware is supposedly not interchangeable.
Does anyone
Title: Getting asterisk-addons installed on Debian?
Hi,
I am having some trouble installing asterisk addons on Debian. I wish to do this to use mysql billing.
I have mysql and mysql-devel packages installed I think!?
pbx01:/usr/src/asterisk-addons# dpkg -l mysql-server libmysqlclient*dev
On Sat, Mar 05, 2005 at 01:19:24PM -, C. Tomlinson wrote:
Hi,
I am having some trouble installing asterisk addons on Debian. I wish to do
this to use mysql billing.
snip
make[1]: gcc: Command not found
You need a C compiler, try apt-get install build-essential
Hope this helps,
--
Wolfgang S. Rupprecht wrote:
FYI: FWD shows a different inbound prefix:
**164 e164.org8781039311
Yes that works as well, and was issued by another company, the contact
at FWD asked if we could route that to them as well, I prefer the other
range because it's shorter and
Hi ml, this is my problem:
I have an Asterisk on remote site (my office) and two x-lite at home behind
a ful cone nat. Both my ua can register, I can place and receive calls from
both the phones and I can hear voice, so I don't think I have nat problem
but when when i place a call if the called
Hi,
I've just compiled and installed Asterisk (1.0.5). After some problems
with codecs I could successfully connect to server by:
[EMAIL PROTECTED]
Next I created account at iaxtel.com and configured iaxcomm to work with
this account. Unfortunately after that I had problem with logging as
I am able to transfer a call to call parking using '#' without any
problems since I have the 't' option in my Dial() line. However, if no
one picks up the call after it has been parked and a timeout occurs
(such that the call is returned back to the original extension), the
call is no longer
Hi,
Thanks. I idd that and now get different errors:
pbx01:/usr/src/asterisk-addons# make install
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c`
app_addon_sql_mysql.c:15:27: asterisk/file.h: No such file or directory
app_addon_sql_mysql.c:16:29: asterisk/logger.h: No
Hi-
I am attempting to setup my Budgettone phone for
use with my * server and am having problems obtaining an IP address. I have
checked the phones settings to
make sure it has dhcp enabled and it is. The
display says no IP. I bought the phone but do not have any documentation other
than
What could be preventing the phone from picking up an IP address?
Do you have a DHCP server on your network?
Stuart
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To
Hi. I am trying to set up my Asterisk box to block anonymous calls.
I am having some grief from telemarketer calls to my number and I would like
to block it.
I see from my CDR's that some of my callers also have unknown in their
FROM field. I would like to let them through. Only block the FROM
ztcfg - works, too.. after a timing source change... power cycle
works.
-
Yeah, we rocked the vote all right. Those little
bastards betrayed us again.
- Hunter S. Thompson on the 2004 election.
On Fri, 4 Mar 2005, Steven Critchfield wrote:
Grab a copy of the Asterisk source, and untar it into /usr/src. Once
you've done this, make sure that files such as
/usr/src/asterisk/include/asterisk/file.h exist.
Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
+44 (0)7870 699 479
http://integrics.com/
C.
Stuart Ford wrote:
What could be preventing the phone from picking up an IP address?
Do you have a DHCP server on your network?
Stuart
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There is a partnership between Elmeg and snom. We are using their
plastic (in the snom 190/200/220), they are using our hard- and software
(in the Elmeg 290). Elmeg have a long experience in making phones and we
have experience in making hard- and software for VoIP (as long as it can
be in the
I should have added that you'll be OK with your current build; you'd
have to install the source for Asterisk if you went for a default Debian
anyway.
Alistair Cunningham,
Integrics Ltd,
Telephony, Database, Unix consulting worldwide
+44 (0)7870 699 479
http://integrics.com/
Alistair Cunningham
Asterisk has the ability to do agent queueing and some general ACD
functionality. The functionality doesn't come close to the
functionality/flexibility of Avaya's Expert Agent functionality, but *
won't cost you several hundred thousand dollars for deployment either.
As far as reporting, there
Replying to my own post :-(
Yes, I'm top-posting, because no one ever seems to reply to my posts
anyway, I don't want to make you re-read my old post just to find out
what I'm adding.
I have _not_ solved the problem, but I reverted briefly to 1.0.3, and
I can indeed call to FWD without any
Mike are you able to log into the phones
web server configurator page at all? (Im assuming not if it isnt
picking up an ip address).
Are you able to assign an ip address via
the keypad?
Are you able to reset the
handset via the mac code/reset command.
Cheers,
Dean
On Sat, 2005-03-05 at 17:25 +0700, Nattapong Mongkolnavin wrote:
I have a problem using AGI cmd SAY DIGITS. For some reason I cannot
here any thing when the script got executed. However if I use the cmd
SAY NUMBER I can here * reading the number fine.
fputs($stdout, SAY DIGITS
Thought I had this fixed, but it turns out it is not. I've been
wracking my brain. Here is what I have done:
- Tried 3 different Qwest PSTN lines (just in case it was a line issue)
- Tried calling same number from an analog phone plugged directly into
the Qwest line - NO PROBLEMS.
-
It's nice to see that some people think so highly of
themselves and are above all others. It's quite
amusing to watch people like you give thinking so
highly of yourself and so little of others. In the
spirit of Asterisk and Mark's organization-Digium, I
certainly could understand why you aren't
Good success story.. I'll keep in mind that router just in case.
Thx David.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J Carter
Sent: Sábado, 05 de Marzo de 2005 04:18 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On March 5, 2005 08:14 am, Androtech wrote:
I bought one Trust 56k V92 PCI Internal Modem MD-1100 which has the 1057
Motorola Chip, and I installed it on my linux box.
When I try to load the module wcfxo, I cannot load it (zaptel is already
loaded):
Not to rub salt in the wound, but do you
I remember reading somewhare that you should disable as many unused codecs
on sip phones as possible to reduce bandwidth.
I'm not sure what I have to do for each type of device I'm using.
I have 2 sipura-2000's and 2 Grandstream BudgeTone 100's and a
sipura-3000, and a few Xlite PC clients.
On March 5, 2005 10:48 am, asterisk phones wrote:
It's nice to see that some people think so highly of
themselves and are above all others. It's quite
amusing to watch people like you give thinking so
highly of yourself and so little of others. In the
spirit of Asterisk and Mark's
when you use Dial application without tTm options, and two User agent
use same g.720 codec,
The two User agent will transfer media with passthrough.
You will no need to install g.729 codec module
If you want some commerical G.729 codec, pls visit,
http://www.voip-info.org/wiki-ITU+G.729
On
On Sat, 2005-03-05 at 11:01 -0500, Andrew Kohlsmith wrote:
On March 5, 2005 08:14 am, Androtech wrote:
I bought one Trust 56k V92 PCI Internal Modem MD-1100 which has the 1057
Motorola Chip, and I installed it on my linux box.
When I try to load the module wcfxo, I cannot load it (zaptel
Andrew Kohlsmith wrote:
On March 5, 2005 08:14 am, Androtech wrote:
I bought one "Trust 56k V92 PCI Internal Modem MD-1100" which has the 1057 Motorola Chip, and I installed it on my linux box.
When I try to load the module wcfxo, I cannot load it (zaptel is
Currently I have one server running Fedora Core 3 AMD 64bits (on a 3mbits DSL with 640kbits upload) and the second server is running on Mac OS X (on a 512kbits SDSL) I'll change it soon for a PC with Fedora Core 3 but I know the G.729 isn't available for Mac OS X.
Is there another codec that do
Today, We have added INVITE Authentication. This seems to bring a large
amount of problems to people in the way since they can't make outbound
calls. Here's what needs to be done. You need to add three variables to
your peers or friends, username, authuser, and secret.
username=phonenumber
Has anyone done Voice Over Frame Relay with Asterisk.
With Frame Relay work reliably with Asterisk? Any
experiences?
If you're talking about transporting voip calls across a path that
includes frame relay links, yes it works just fine if you frame
network is not congested.
Frame relay
Ummm... This isn't a Digium Run list. This is a Digium sponsored list. My
understanding (someone please correct me if I'm wrong) is that this list *is
not* a Digium support list. This list is a forum for Asterisk discussion by
users. As such, I would suggest that all topics of discussion
I'm looking into solutions for providing a live stream of an event in
Belgium [1] - for example, as follows:
* Event -- mobile phone -- software answering machine -- Internet
server
* Event -- mobile phone -- VOIP -- Internet server
The live stream should be available in a format so that
Thank you Wiley.
I guess I had the "q" version installed. I
removed everything and tried with "r" and followed the wiki instructions. Simple
Answer-MusicOnHold works fine, so I guess my problem is resolved.
- Original Message -
From:
Wiley
Siler
To: Asterisk Users
Currently I have one server running Fedora Core 3 AMD 64bits (on a 3mbits DSL
with 640kbits
upload) and the second server is running on Mac OS X (on a 512kbits SDSL) I'll
change it soon
for a PC with Fedora Core 3 but I know the G.729 isn't available for Mac OS X.
Is there another codec
Something like this sis similar to what you are looking for I think.
C
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Felix E. Klee
Sent: 05 March 2005 17:17
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk for Live-Stream?
I'm
As far as I can make out the root password for the ISO download is
supposed to be epping or EPPING depending upon which version you are
using.
I've downloaded an ISO image from the following link but neither passwords
seem to work :(
Ummm... This isn't a Digium Run list. This is a Digium sponsored list. My
understanding (someone please correct me if I'm wrong) is that this list *is
not* a Digium support list. This list is a forum for Asterisk discussion by
users. As such, I would suggest that all topics of
Great, thanks, that was the information I was looking
for.
--- Rich Adamson [EMAIL PROTECTED] wrote:
Has anyone done Voice Over Frame Relay with
Asterisk.
With Frame Relay work reliably with Asterisk? Any
experiences?
If you're talking about transporting voip calls
across a path
Rich Adamson wrote:
snip
There seems to be about a half dozen self-appointed list cops, and none of
them speak for Mark, digium or asterisk. Several of those are lurking on this list only
to find fresh meat to sell their services to.
It's obvious who they are.
Indeed.
Much easier to config
snip
I've downloaded an ISO image from the following link but
neither passwords
seem to work :(
http://ovh.dl.sourceforge.net:80/sourceforge/asteriskathome/as
teriskathome-0.6.iso
any one know the password for this one?
--
Regards
Phil
The root password for 0.6 is (I
That worked a treat many thanks.
--
Regards
Phil
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Andrew Kohlsmith wrote:
On March 5, 2005 08:14 am, Androtech wrote:
I bought one Trust 56k V92 PCI Internal Modem MD-1100 which has the 1057
Motorola Chip, and I installed it on my linux box.
When I try to load the module wcfxo, I cannot load it (zaptel is already
loaded):
Not to
On March 5, 2005 12:22 pm, Rich Adamson wrote:
There seems to be about a half dozen self-appointed list cops, and
none of them speak for Mark, digium or asterisk. Several of those are
lurking on this list only to find fresh meat to sell their services to.
It's obvious who they are.
I have
On March 5, 2005 11:01 am, Andrew Kohlsmith wrote:
specifically these types of problems. Maybe someone else on this list is
more forgiving than I am but I really hope not.
I apologize for this remark. I still do feel, though, that if you're this new
to asterisk that you should have purchased
Hi,
I have two Asterisk servers and I forward calls from one to the other.
How do I reload extensions included in a switch statement in
extensions.con? I have tried extensions reload, reload and restart
now, and it's only restart now that works. Is this how it is supposed
to work or can it be
The code is configured to allow use of either mysql or postgres, so you
will
need to install the postgres-dev package, or comment out all postgres
related
code.
Once you have the postgres libraries installed you have two more changes
to make.
line 645 needs to become:
On March 5, 2005 11:57 am, Dave Cotton wrote:
Just a question, where's the Dev Lite Kit on Digium's site?
I meant Dev Kit and you're right, it's an FXO and FXS on the carrier card.
US$195.
The PCI Dev kit would give him an FXO and an FXS which may be more than
some people want, perhaps the
Hi,
I've no experience with the TE110, but this is a known problem with the
TE405 and TE410. They apparently can get locked up, and only a power
cycle will clear it.
Good hint, I'll take that into account when testing.
Hope the TE100 is better built than that, though. At least, for
On March 5, 2005 12:02 pm, John Novack wrote:
Since Digium no longer suppliers this card, they were denied NOTHING!
They offer comparable hardware. TDM410P is $113.
There is a LOT of traffic on this list about products that are not
supplied by Digium. Do you want to exclude those also?
The
Hi!
I'm looking into solutions for providing a live stream of an event in
Belgium [1] - for example, as follows:
How about icecast:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Ices
Another approach:
Dial into a MeetMe conference, and connect some client to that conference
- Original Message -
There is a LOT of traffic on this list about products that are not
supplied by Digium. Do you want to exclude those also?
The Sangoma guys typically handle support for their own product, even on
this
list. Atacomm's card hasn't hit the market yet. The Sipura people
And if nobody's going to educate the newbies, then how will they ever learn?
Do you believe in letting your children do whatever they want, too? There
are 'defacto' rules for any system. No, I don't have my shiny ListCoptm
There is a difference when you are that childs father or mother but
Hi Nathan,
Nathan C. Smith wrote:
I'm running asterisk stable 1.0.5 and I'm trying to get the netweb eezee
phone version v1.37.008 to talk IAX to asterisk. The pages I saw in the
Try the wiki, myself and someone else wrote up a pretty big howto and
tips and tricks on these phones.
Dnia 2005-03-05 15:04, Uytkownik Marcin Zajczkowski napisa:
I've just compiled and installed Asterisk (1.0.5). After some problems
with codecs I could successfully connect to server by:
[EMAIL PROTECTED]
Next I created account at iaxtel.com and configured iaxcomm to work with
this account.
On Sat, 2005-03-05 at 19:30 +, Mike Dent wrote:
And if nobody's going to educate the newbies, then how will they ever learn?
Do you believe in letting your children do whatever they want, too? There
are 'defacto' rules for any system. No, I don't have my shiny ListCoptm
There is
Mike Dent wrote:
And if nobody's going to educate the newbies, then how will they ever learn?
Do you believe in letting your children do whatever they want, too? There
are 'defacto' rules for any system. No, I don't have my shiny ListCoptm
There is a difference when you are that childs
What iax2 softphones are you guys using?
Ive trying some but I find some lack certain features and others have them
but lacks others.
For example, I tried firefly, simple interface but seems it can only handle
1 line, no MWI.
IAX Phone has multiple lines and MWI but seems it can only handle
Hello! I am a newbie with asterisk, I´d like to install capi on FC3, I´ve
tried to follow a little howto
(http://voip-info.org/wiki-Asterisk+Linux+Fedora), but it is for FC1, and
when I do a modprobe fcpci it fails (module not found).
Please some help!!
Hi list!
I'm using phones that emulate a Cisco 7940 with chan_sccp. When I was
using Asterisk 1.0.5 (bristuffed) I never had any such message on the
console.
The phones do work.
Is this a bug in chan_sccp or a feature of asterisk 1.0.6?
Thx!
___
On March 5, 2005 02:30 pm, Mike Dent wrote:
There is a difference when you are that childs father or mother but
you are neither?
You have a point, but... (read on)
Do you stop people in the street when you see them doing things wrong
and try and tell them you shouldn't be doing it like that,
On March 5, 2005 02:46 pm, Paul wrote:
Maybe one of the free web-based forum packages will eventually offer an
elitist or impatient mode. Before you can post, you do the required
reading and pass online exams. The idea is to weed out people who think
README is just another geek buzzword. You
Well, considering I'm on topic, I shouldn't get flamed to badly for this. I
have a bunch of these working well in my home experiments:
http://www.laptops4me.com/product_info.php/products_id/1444
And yes that price is correct and they do arrive. :)
Not everyone can justify buying the supported
On 5 Mar 2005, at 18:44, Alfredo Sola wrote:
Hi,
I've no experience with the TE110, but this is a known problem with
the TE405 and TE410. They apparently can get locked up, and only a
power cycle will clear it.
Good hint, I'll take that into account when testing.
Hope the TE100 is
Anyone using this Sip phone with Asterisk?
If you have had success getting the message waiting indication to work,
please contact me off list.
TIA
John Novack
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On March 5, 2005 04:15 pm, tim panton wrote:
My feeling (unsupported) is that the powercycle does a better job of
forcing the far end
of an E1 (e.g. the PTT's equipment) to start afresh than just
reinitializing the cards.
If you turn the power off you can be sure that you are going to drop
As far as I can make out the root password for the ISO download is
supposed to be epping or EPPING depending upon which version you are
using.
I've downloaded an ISO image from the following link but neither passwords
seem to work :(
Does any have ... or know where I can find firmware to convert a
DVG-1120M (MGCP) to a DVG-1120S (SIP)??
Thanks,
Rob
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Can anyone recommend a Digium Reseller in the UK ?
Thanks in Advance
Nigel
begin:vcard
fn:Nigel Taylor
n:Taylor;Nigel
org:ITAzure Limited
adr:15 Warren Park Way;;Dunn House;Enderby;Leicestershire;LE19 4SA;United Kingdom
email;internet:[EMAIL PROTECTED]
title:Technology Director
tel;work:0116 286
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
I think where the problem comes in is that people take this forum to be
asterisk-biz half the time. I need X done **RIGHT NOW**!! I DEMAND
HELP!!
-- take it to -biz, there are dozens if not hundreds of consultants who
will
Thanks for this info, Dan! I noticed immediately that outbound was broken, and
inbound was OK. I saw your posting just prior to going berserk... a warning
email from Broadvoice would have been nice, they knew to email me when that
SIP-patch from edvina came out some months ago.
Anyway,
I don't know if this is still true, but Iax clients had problems when you
check them with qualify (set latter to no)...
HTH,
Rob.
- Original Message -
From: Anton Krall [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent:
Fredrik wrote:
I see from my CDR's that some of my callers also have unknown in
their FROM field. I would like to let them through. Only block the
FROM anonymous that the telemarketers use.
Fredrik, I found something on the Wiki a while back... Try this...
exten = s,1,Answer
exten =
Andrew Kohlsmith wrote:
On March 5, 2005 02:46 pm, Paul wrote:
Maybe one of the free web-based forum packages will eventually offer an
elitist or impatient mode. Before you can post, you do the required
reading and pass online exams. The idea is to weed out people who think
README is just
Can anyone recommend a source of Digium hardware in the UK ?
Thanks in advance
Nigel
begin:vcard
fn:Nigel Taylor
n:Taylor;Nigel
org:ITAzure Limited
adr:15 Warren Park Way;;Dunn House;Enderby;Leicestershire;LE19 4SA;United Kingdom
email;internet:[EMAIL PROTECTED]
title:Technology Director
An Open letter to Broadvoice from an Asterisk user...
(This is not a solicitation for support from the Asterisk list. The
specifics of my problems have already been emailed to their support
team.)
May I suggest:
1) Updating your website that tells how to configure Asterisk for Broadvoice.
2)
Anyone knows what are the variables in an inbound IAX2 call who
reflect the actual codec and DNID, DNIS, original peer description, I'm only
able to see it during an iax debug
Timestamp: 3ms SCall: 1 DCall: 0 [XX.XX.XX.XX:5036]
VERSION : 2
CALLED NUMBER :
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