Hi,
My asterisk sometimes stop responding to iax calls.
In the log, I've found this:
Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for
'IAX2/trunkjstpcn-3'
Mar 29 13:35:45 DEBUG[13002] chan_sip.c: update_call_counter(1409) -
decrement call limit counter
Mar 29 13:35:45
Hi,
I'm trying to call an extension and then transfer the call
to another extension, but something strange happens.
This is the extension:
exten = _9.,1,Dial(CAPI/ISDN4/${EXTEN:1}/b,tT)
When I dial any number starting with 9, I always
get CALL FAILED, but the called party still receive
the call
Hi,
just to complete this thread if someone faces a similiar problem:
The missing DID is caused by our telco company. It only happens when
having two different ptp lines (with different numberblocks) and calling
from one of these to the other. Calls from any other line in the world
come in
Or just pop down to your local computer store and get a molex splitter.
Regards,
Mark.
Rich Adamson wrote:
The fxs ports have to generate ringing voltage (about 90 vac) and they
use the 12 volt power supply to do that. When an fxs port is not
ringing, it consumes about the same amount of
That implies that the 2850 has a standard molex connector anywhere inside of
it, which is not the case.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Davies
Sent: Wednesday, March 29, 2006 6:34 AM
To: [EMAIL PROTECTED]; Asterisk
Dmitry,
it seems to me that just in the definition of the extension, (Outbond CID - thru the AMP) you just define the CID of that extension.
be carefull to give the proper CID, within your block, to that extension.
good luck,
Mickey
On 3/29/06, Dmitry Ivanov [EMAIL PROTECTED] wrote:
On Tuesday
You just add the same agent to both queues (don't use groups), like in
queues.conf:
[queue1]
member=Agent/101
[queue2]
...
member=Agent/101
Now Agent 101 is a member of both queues, and will not be called while
s/he is on conversation.
l.
On Wed, 29 Mar 2006 09:02:11 +0200,
We tried to give a MAX4000 behind a Asterisk with TE 405, but the connection is very slow (max of 28.8) and we have also a problem with a Fax Server behind the Asterisk, We loose lines and so on.Did anyonehave an idea?ThanksNico--
-Ursprüngliche Nachricht-Von: Don Pobanz [EMAIL
Ok,
Understood all this.. but isn't that for making 'static' agents? What
if I want my agents to be able to log in/out of the queues... ie when
they are not here.
On 3/29/06, Lenz [EMAIL PROTECTED] wrote:
You just add the same agent to both queues (don't use groups), like in
queues.conf:
Is there anyway in asterisk to figure out how much time an agent has
spent on the phone? I know I can see total time for a call (inbound
or outbound) but where/how do I view queue stats?
___
--Bandwidth and Colocation provided by Easynews.com --
On 03/29/06 13:06 Andres said the following:
It works perfectly with other values we have tested of 40 and 60. We
currently use 60 on all our servers. It cuts down on bandwidth for a
G279 call to about 15Kbps.
with 60ms packets, is a packet loss or two noticable ?
--
Regards,
Hi all,
I did not get this error in Asterisk 1.2.5 release. I am testing on Asterisk SVN-trunk-r15187 to avail the PARKEDAT variable.
- I park the call using ParkAndAnnounce
- plays moh.
- accept the call using ParkedCall
The following errors are coming on the console and there is oneway audio
With such a configuration, agents get to be available (on all queues) as
soon as they log in and stop to be availbl when they log out. That's why
youu use agents instead of, say, SIP/123.
The other alternative is to have an agent join each queue dynamically via
AddQueueMember() and then log
Hello,
(I have asked it some time ago in Asterisk-es mailing list, so excuse me if
anybody receive it twice.)
I am trying to send SMS in Spain using landlines. It seems that
app_sms.c only handles Protocol 1, but Spain and Italy are using
Protocol 2.
I have been searching in Internet without
I know Vonage doesnt officially have a bring
your own device type program, but they do offer a softphone. Has anyone
gotten Asterisk to connect directly to Vonage? This would be a great help!!
___
--Bandwidth and Colocation provided by
The reality is, of course, that telephone systems have provided this
function for many years. A DSS/BLF is available on MANY so called legacy
systems, so until this function is readily available , customers that
require a receptionist will continue to go elsewhere.
Perhaps it is time to rethink
LOL Yes it does.
Ok.. so you are saying I should put in agent numbers in the queues
rather then phone extensions? I'm having serious problems with the
agent joining queue... for some reason asterisk is letting me log
agents in that aren't configured in agents.conf and then shows them
logged into
Brian Deep posted this to the list back in August. I still haven't tried it
myself but he said it worked. If you try it out and it works, please post back
your success. (or failure if that is the case)
Jason
sip.conf:
[general]
externip=X.X.X.X
port=5060
bindaddr=X.X.X.X
On 29/03/06, Steve Jones [EMAIL PROTECTED] wrote:
I know Vonage doesn't officially have a bring your own device type
program, but they do offer a softphone. Has anyone gotten Asterisk to
connect directly to Vonage? This would be a great help!!
I'm not a Vonage customer, but I did spot
Plus see this:
http://www.voip-info.org/wiki/view/Asterisk+and+Vonage
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Bowyer
Sent: Wednesday, March 29, 2006 8:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On the same line
I see we can not connect Asterisk to Vonage. But has anyone tried
unlocking the Motorola ATA they send you? I would like to configure it so
that I can connect to my office Asterisk installation.
Thank you,
Jyran Glucky
The owner of my company just asked me for an Asterisk brochure. Has
anyone seen such a creature? I know of some really informative
websites, but I think a pdf would be priceless at this point.
Thanks,
Bob McDowell
EMAIL PRIVELEGED CONFIDENTIAL CLIENT COMMUNICATION
*** PRIVILEGED
I have this working. I have Asterisk connecting to my Vonage Linksys device
via Digium Wildcard X100P. No magic needed ;)
Curt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell
Sent: Wednesday, March 29, 2006 9:25 AM
To: [EMAIL PROTECTED];
On 3/27/06, Erick Perez [EMAIL PROTECTED] wrote:
Hi,
Our asterisk installation will be a man-in-the-middle providing
local,long,international VOIP services to our customers and our asterisk
will be connect via VOIP to international carriers.
We use asterisk 1.2.5 with mysql in centos 4.2
snip wonderful place to start. Nothing against Asterisk or Linux. My build fromscratch issues are only due to my lack of Linux experience.../snip the only way to learn is by playing. a little over a year ago i knew nothing about linux. google. is your friend.
New Yahoo! Messenger with Voice.
Bob McDowell wrote:
The owner of my company just asked me for an Asterisk brochure. Has
anyone seen such a creature? I know of some really informative
websites, but I think a pdf would be priceless at this point.
Bob,
Check on Digium's website. I know there is such a creature there.
Steve Kennedy wrote:
On Tue, Mar 28, 2006 at 07:40:06PM +, Tofik Suleymanov wrote:
Each of the two lines have their own entry in sip.conf and i can see
each line registered in 'sip show peers'.
I can dial each line from outside successfully but when one line is busy
i can't reach the
Vahan Yerkanian wrote:
Tofik Suleymanov wrote:
1. assume 1-st line is in use
2. after dialing 2-nd line from outside i immediately go to the
voicemail announcement (also i immediately go to voicemail if i dial
from extension to extension both of which are on the same sipura device)
Check
Dear All,
I have the following setup;
SER/External Asterisk -- Firwall -- Internal Asterisk -VPN- Users
At the moment;
Anybody can register with our SER proxy and call each other using VoIP.
Anybody can call one of our internal users via our SER/Asterisk gateway.
The INVITE is sent to our
I did. I will do so again, just to be sure. I am one of those 'search
first' type of list guys, and would not want to waste this list's time
if I could help it...
Bob McDowell
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darrick
Hartman
Sent:
Digium.com has pdf brochures on Asterisk and their hardware you can
download.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob McDowell
Sent: Wednesday, March 29, 2006 8:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Capatres released some time ago a solution with an ITSP.
Maybe it could help
http://blogs.capatres.com/index.php?op=ViewArticlearticleId=18blogId=1
Carles Pina i Estany escribió:
Hello,
(I have asked it some time ago in Asterisk-es mailing list, so excuse me if
anybody receive it twice.)
Hi all
I am setting up a asterisk
iam able to setup a outbound calling its wrking
still some tweaking required iam doing
now i got some DID routing to my IP from my provider
i want them to give to another Client from my network
or out side my network using my asterisk
i am not looking forward
Actually, they do have a bring your own device
program. It's called Business Plus. Works great with
Asterisk.
http://www.thevoipconnection.com/store/catalog/product_16220_Vonage_Business_Plus.html
Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728
ext. 611 sip:[EMAIL
Hello all!
Ive got a problem with the IAX setup. Im
previously only experienced with SIP, so that may be part of the problem. However,
Ive managed to register with the IAX server without any trouble
(register line apparently works as it should), and I am also ale to make
outbound calls.
Hi,
is it possible to force FOP to reload its configuration files
(op_buttons.cfg and op_style.cfg) while it is working? I tried to click
on the refresh icon but nothing happens.
TIA
Giorgio Incantalupo
___
--Bandwidth and Colocation provided by
Hello-
I have a low traffic AAH setup, a few hardphones, a few softphones, 50 calls
per day max. I used the AMP Digital Receptionist to
make a simple voice menu: Thank you for calling . I did this for both
Normal times and After Hours times. It worked fine.
I then went to the AMP
Hello,
from where do you see errors which are generated by tftp. I have
searched from Google and I didn't find that. Maybe you can help me. I
have FC4 too :)On 3/17/06, Joseph Rothstein [EMAIL PROTECTED] wrote:
Greetings to all.I am hoping someone can help me out with a problem I am having getting
Because we still have to live with - the current - seven only
limitation.
We have found that training generally results in folks understanding
how to use a PBX vs a key system
For many years I installed large PBX systems, Rolms, ATT, Nortel, etc
and often no blf/dss was available, or
Telefónica use both protocols to deliver an SMS (UBS1 and UBS2).
Most nowadays fixed-devices (in Spain) are UBS2 but there are UBS1 too.
The Telefónica messaging platform have the information of terminals of each
subscriber and its access protocol.
good luck, hope it helps!!
Fran
-Mensaje
I should add, I have now managed to get a
little further, but I still get an error message:
Mar 29 18:17:34 NOTICE[11943]:
chan_iax2.c:7213 socket_read: Rejected connect attempt from 213.160.242.5,
request '[EMAIL PROTECTED]' does not exist
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno:
is it possible to force FOP to reload its configuration files
(op_buttons.cfg and op_style.cfg) while it is working? I tried to click
on the refresh icon but nothing happens.
No, you have to kill the op_server app and restart it
___
--Bandwidth and
If you are using the safe_opserver daemon, 'killall op_server.pl' works
fine. You have to kill the op_server process to force a reread of the .cfg
files
-Original Message-
From: Giorgio Incantalupo [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 29, 2006 9:02 AM
To: Asterisk Users
You made some change to something using AMP and it overwrote the
extensions_additional.conf file as it was designed to do. The only safe
place to put customizations is in extensions_custom.conf.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL
I'm currently using Asterisk running version 1.2.5 and trying to use cdr_odbc to connect to a Microsoft SQL database. I have everything running, but the insert statement being sent to database doesn't appear to have the start, answer, end information in it.
Below is the insert statement that MS
On Wed, 29 Mar 2006 11:05:05 -0500, Agur Koort [EMAIL PROTECTED] wrote:
Hello,
from where do you see errors which are generated by tftp. I have searched
from Google and I didn't find that. Maybe you can help me. I have FC4
too :)
You do a tail on /var/log/messages
By the way other dude,
Hello,
I have tried to install this phone for hours now and I can't get it
working. Maybe someone can help me :) I have searched for more info
from everywhere but there isn't much about 7910 :(
>From the CLI I get this:
NAME
ADDRESS
MAC
Reg. State
===
Using this same method would I be able to add a cutsom menu in astcc (like
call recording), by having it drop back into the IVR and then back to the
agi?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JP Carballo
Sent: Wednesday, March 29, 2006 1:52 AM
Just noted that in indications.conf.sample in SVN that in the section
[fr] the variable callwaiting is written as callwait. Could someone add
the ing?
Thanks.
--
Dave Cotton [EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
My question is: how can I set specific caller id for outgoing
PRI calls?
Here in Italy I have a E1 PRI line with DID: +39 local-zone-prefix
did-block-prefixdid-block-ext
I was able to set CallerIDnum only after some attempts: I had to set it only
to:
did-block-prefixdid-block-ext
without using
AAH uses a database to store the configs.
It then outputs the database info into the text files for asterisk to use.
ALL _additional files are built from the database and hand edits WILL be lost.
If you are trying to do something that can't be done in the web interface, you
need to put that into
Hi all,
I would like to hear from you, SjPhone has the option to Do not Send
silence (audio options, advanced), should i use this or remove this
option?
Everything ran well until now, but there was few people on my server,
i'm increasing sip extensions and i want to avoid complains from the
Because the VoiceBlue is only 4 channels and I am supporting 100 cell users,
I needed a way to overflow calls to the PRI of all 4 channels are full.
Unfortunately, there seems to be no built-in mechanism to determine if the
gateway is full, so this script parses the output of asterisk -rx sip show
Giorgio Incantalupo wrote:
Hi,
is it possible to force FOP to reload its configuration files
(op_buttons.cfg and op_style.cfg) while it is working? I tried to
click on the refresh icon but nothing happens.
killall -HUP op_server.pl
Doug
--
Ben Franklin quote:
Those who would give up
Wilson Pickett wrote:
is it possible to force FOP to reload its configuration files
(op_buttons.cfg and op_style.cfg) while it is working? I tried to click
on the refresh icon but nothing happens.
No, you have to kill the op_server app and restart it
This is really the day for new experiences
sorry for the load on the mailing list, but this will be the last issue
I try to solve before I take the night off;)
So Ive got the incoming calls to
work with a not-so-well solution (could therefore still need some feedback on
the previous
Hi all
I don't manage to get asterisk 1.2.5 or 1.2.6 running with the zaphfc
driver
The scripts from junghanns.net do download a very old libpri and asterisk
version which is too buggy for me to use.
Isn't there an acutal patch to get zaphfc support in *?
-Benoit-
I have a Digium TDM400P card with 1 FXS and 1 FXO module running on my
FreeBSD 6.0 server.
While I am on the road, I would like to save on costs by using a soft-phone
from my laptop to call in to a telephone connected to this card.
I installed both Asterisk and Zaptel drivers from the ports, but
On 3/29/06, Benoit Panizzon [EMAIL PROTECTED] wrote:
Isn't there an acutal patch to get zaphfc support in *?
You even have 3 possible ways out..
1; you stay with the current bristuff (a somewhat older
zaptel+asterisk, but is this really making a difference?)
2; you use a visdn snapshot
Jeremy wrote:
Using this same method would I be able to add a cutsom menu in astcc (like
call recording), by having it drop back into the IVR and then back to the
agi?
Of course.
It's just a matter of setting things up in the Dialplan before passing
the caller to astcc.
In my IVR for
There is a proble to put an H323 Asterisk server behind an iptables firewall?
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Are both protocols enabled? I remember I had to first send an SMS with
the Domo (an analog phone with sms capabilities) before I could even
receive them.
Maybe protocol 1, even if it's implemented, needs to be enabled someway.
Julian J. M.
On 3/29/06, Fran [EMAIL PROTECTED] wrote:
Telefónica
If you open h323 port and rtp ports, it should work.
Il Neofita escribió:
There is a proble to put an H323 Asterisk server behind an iptables
firewall?
___
--Bandwidth and
I am switching from IAX2 to SIP for my inter-Asterisk transport due to
assorted quality issues following the 1.2.4 upgrade.
On the server that SENDS the call, I have the following in SIP.CONF:
[192.168.1.2_OB]
type=peer
fromuser=OB
host=192.168.1.2
And in EXTENSIONS.CONF
exten =
Shad Mortazavi wrote:
What I would like to do is to redirect external SIP calls to our
external Asterisk server. e.g if I call sip:[EMAIL PROTECTED] I would
like the call to be routed from our Internal Asterisk server to our
External Asterisk server via IAX2 and for the external asterisk
Hi,
Did anyone hear about a recent ruling which makes it illegal to have
caller-id set to anything except what is on the account of the user?
IE... If your name is Joe Smith you can't have Mary Smith set as
the caller-id name, unless mary smith is also on your account.
I believe that they covered this exact procedures at www.voip-info.org.
Look for the topic on connecting two Asterisk servers. They outline three
different ways that you can do so.
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Yes, it will work.
On 3/29/06, Kiffin Gish [EMAIL PROTECTED] wrote:
I have a Digium TDM400P card with 1 FXS and 1 FXO module running on myFreeBSD 6.0 server.While I am on the road, I would like to save on costs by using a soft-phone
from my laptop to call in to a telephone connected to this card.I
DId you hear about the law that says you can't drive over the posted speed? How about the one about junk faxes, or the one about spam?
Seriously, I'd like to see how this will be enforced.
While I am sure this is to combat telemarketers faking their caller ID, I'm sure they already have some
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Sent: Wednesday, March 29, 2006 10:43 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: AAH lost my IVR phrases
AAH uses a database to store the configs.
It then
The h323 channels doesn't have any support for NAT. You'd need to
register with a properly configured gnugk for that.
Julian J. M.
On 3/29/06, Alberto Sagredo [EMAIL PROTECTED] wrote:
If you open h323 port and rtp ports, it should work.
Il Neofita escribió:
There is a proble to put an H323
Dinesh Nair wrote:
On 03/29/06 13:06 Andres said the following:
It works perfectly with other values we have tested of 40 and 60. We
currently use 60 on all our servers. It cuts down on bandwidth for a
G279 call to about 15Kbps.
with 60ms packets, is a packet loss or two noticable ?
On 10:08, Wed 29 Mar 06, Colin Anderson wrote:
Because the VoiceBlue is only 4 channels and I am supporting 100 cell users,
I needed a way to overflow calls to the PRI of all 4 channels are full.
Unfortunately, there seems to be no built-in mechanism to determine if the
gateway is full, so
On Wed, 29 Mar 2006 14:19:41 -0500, Matt [EMAIL PROTECTED] wrote:
Did anyone hear about a recent ruling which makes it illegal to have
caller-id set to anything except what is on the account of the user?
Where did you hear this? Can you give a link? Looks like I'm going to
jail, tee hee.
This is not an issue with asterisk.
Asterisk does not use the database, nor has a web interface.
AMP (included with AAH) is an addon that uses a database for it's configs.
The only way that asterisk can use them, is to write them out to standard
asterisk config files.
The rules with AMP, (which
On 29/03/06, Matt [EMAIL PROTECTED] wrote:
Hi,
Did anyone hear about a recent ruling which makes it illegal to have
caller-id set to anything except what is on the account of the user?
A ruling in what jurisdiction?
Peter
--
Peter Bowyer
Email: [EMAIL PROTECTED]
I was told it was more along the lines of preventing stalking, but I'm
sure telemarketers fit in there :)
IE... Mary Smith seperates from Joe Smith... Joe Smith gets phone
service and wants his CID to read 'Jane Smith', who is Mary's sister,
so that Mary will answer when Joe calls.
On 3/29/06,
Hi, i have asterisk 1.2.5 working fine from a database in the following way:
asterisk loaded as root
realtime logging to a mysqld 5.x daemon listening in localhost
database is astbill with username astbilluser allowed to the database
the root user in mysqld has a password
res_mysql.conf in
On Wed, 29 Mar 2006 14:54:19 -0500, Matt [EMAIL PROTECTED] wrote:
IE... Mary Smith seperates from Joe Smith... Joe Smith gets phone
service and wants his CID to read 'Jane Smith', who is Mary's sister,
so that Mary will answer when Joe calls.
I was always under the impression that the telcos
Jim Hanlon wrote:
1. The alterations to the config files made via AMP Setup pages are archived
in the Asterisk DBMS, but changes made via the AMP
Maintenance pages are not (Apparently. It's hard to be sure what the rules are).
This is an [EMAIL PROTECTED] issue: The Setup page is provided
Bjørn O wrote:
In my extensions.conf I’ve got an entry for the phone number that I’m
supposed to receive calls on:
[default]
Exten = 11223344,1,Dial(SIP/1000)
exten =
and not Exten =
--
Daniel
___
--Bandwidth and Colocation provided by
Right, but SOMEONE has to have that number linked to 'Jane Smith'
somewhere. I do not have any sections or numbers or dockets to quote.
What I was told was that Verizon recently had regulation brought down
on them that prohibits them from setting the caller-id on a number to
something or someone
Hi, I have an AsteriskAtHome installation with
two X100p clones. Everything has been apparently fine for 5 weeks of use
or so, but today, I decided to do some tweaking of my echo cancel parameters,
and I realized that all along, one of my cards has been unavailable for
outbound calls for
Bjørn O wrote:
This is really the day for new experiences – sorry for the load on the
mailing list, but this will be the last issue I try to solve before I take
the night off;)
So I’ve got the incoming calls to work with a not-so-well solution (could
therefore still need some feedback on
It's ${EXTEN} NOT {EXTEN}
Mar 29 17:44:18 NOTICE[11502]: chan_iax2.c:6794 socket_read: Rejected
connect attempt from iax.providers.server.net, who was trying to reach
'{EXTEN}@'
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REJECT
Hello
I have decided to use IAX to send simple voice from one end to
Asterisk as IAX is more light weight than SIP.
As IAX does not use RTP for media transfer, we attach the voice
frames with IAX messages (miniframes may be?)
libiax2 has a function called iax_send_voice that accepts following
On 3/29/06, Brian Roy [EMAIL PROTECTED] wrote:
I'm currently using Asterisk running version 1.2.5 and trying to use
cdr_odbc to connect to a Microsoft SQL database. I have everything running,
but the insert statement being sent to database doesn't appear to have the
start, answer, end
Nah, I don't think so, but not heard of it doesn't mean it doesnt
exist, so I'll just wait and see if anyone comes up with some more
info. Meanwhile I will keep spoofing as needed. :)
On 3/29/06, Matt [EMAIL PROTECTED] wrote:
Hi,
Did anyone hear about a recent ruling which makes it illegal to
This question confuses me.
My understanding is that FreePBX is just AMP renamed and AAH comes with AMP setup as the primary way to manage it.
So, is the question realy that the user wants a newer version of AMP (read FreePBX) then the one that comes either with the newest version of AAH or the
I wanted the user interface of FreePBX over what is
provided in the latest version of AAH. I installed the latest
version of AAH and then just installed FreePBX over the top. It went
fantastic and I do like the FreePBX web interface better than the
latestAAH.
Thanks.
From: [EMAIL
On 3/29/06, Matt [EMAIL PROTECTED] wrote:
Hi,
Did anyone hear about a recent ruling which makes it illegal to have
caller-id set to anything except what is on the account of the user?
IE... If your name is Joe Smith you can't have Mary Smith set as
the caller-id name, unless mary
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ok when I modprobe wcte11xp I get the following
message
ZT_CHANCONFIG failed on channel 26: No such device or
address
Any ideas?
Hi Jon,
Did you set the E1/T1 jumper on the card correctly? for UK/Europe it
should be on the E1 position. In
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
[EMAIL PROTECTED] wrote:
My Asterisk doesn't write CDR's to database via ODBC. Plz, anybody help
me to understand, what I am doing wrong.
Asterisk succesfully writes CDR's into the text log.
Database exists, unixODBC installed and configured
From what I have read, people have successfully connected directly to Vonage
when they pay for the softphone option. I believe the softphone option
requires a different phone number. There are setup instructions if you do a
search.
There is a rumor that Vonage may soon allow all accounts to
If there was some type of ruling, its most likely to be a state public
service commission (different names in different states), and not with
the Fed's, etc.
Right, but SOMEONE has to have that number linked to 'Jane Smith'
somewhere. I do not have any sections or numbers or dockets to
I agree that some of these features which are considered quite basic on
legacy phone systems are a major weakness on the Asterisk system.
It seems to me that more time should be put into getting the basics working
nicely rather than all the work going into the whiz bang bells and whistles.
Linky linky:
http://blog.tmcnet.com/blog/tom-keating/vonage/vonage-opens-sip-credentials.
asp
This is pretty cool too: (essentially free cell airtime in the continental
US)
http://nerdvittles.com/index.php?p=124
-Original Message-
From: mustardman29 [mailto:[EMAIL PROTECTED]
Sent:
Hi everyone,
I am fairly new to the idea of VoIP, although I've been reading about it
off and on for the last few years. Now it is starting to look mature
enough to consider implementing it, but there is one thing that I
haven't been able to get a clear answer on...
With Vonage, you are
Hi guys,
I'm setting up asterisk to run with another pbx server. This pbx server
support a feature that allows 2 extensions connect to the same FXS. No I put
asterisk in the middle.
Asterisk receives the call and dial to a SIP/peer.
How the pbx installed support 2 extensions to one fxs... How
Darrick Hartman wrote:
Bob McDowell wrote:
The owner of my company just asked me for an Asterisk brochure. Has
anyone seen such a creature? I know of some really informative
websites, but I think a pdf would be priceless at this point.
Bob,
Check on Digium's website. I know there is such
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