Is there any peformance problems/etc if I set NAT=yes for all clients?
nat=yes causes Asterisk to respond to the *public* source port and IP
address. Therefore, the only time you should ever have a problem is when
the packets should not go to that port/address, which I think is close
to never.
I just downloaded the new astcc and it includes now a new
field in the list of the cards: Brand Great!
How can I use it in the dialplan?
You can't use it in the dialplan, you use it when creating a card.
Nabeel
___
Asterisk-Users mailing list
I would need that a user could choose between two tarriffs,
... I thought that would be great to use Brands for that.
Like I said, a brand is used when you are creating a card. Therefore,
the markup defined by the brand is applied to the card. Simple?
Nabeel
Are there programmable softkeys in the Cisco SIP software?
Instead of lineX_name, etc., you can use, for example:
speed_label6: Pick-up Group
speed_line6: *8
Nabeel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
It seems to me silly to have a T1/E1 card to connect to a
channel bank when you could just have a 24/30 way FXS card in the
slot in the first place.
Wouldn't a SIP channel bank be better - something that has multiple
FXS and FXO ports but hooks up to Ethernet. I know Wasam (ala Farfon) is
try
I am pretty sure that there are no IAX providers that offer
CallerID name but wanted to double check with the list in
case something has changed recently. Is anyone aware of an
IAX provider that offers incoming CallerID name?
Xetricom Networks, who only have Toronto DIDs, do provide
If you're going to promote your product, you might consider
making sure your web site is up, before giving out the URL.
www.servers-r-us.com
Speaking of website being down, I get the following error when trying to
check prices on your website: Source data is temporarily unavailable.
Please
What happens if a SIP call is routed through more
than one * server?
If canreinvite=yes for all the peers involved, and t or T is not used in
the Dial command, then the audio would get routed directly between the
endpoints.
Also, when setting up an inter asterisk exchange, is all the
data
Is it possible to have only one SIP account that is shared by
several users ? I am currently setting up one asterisk box
for a small company (around 7 users). Can all of them make
simultaneous call using only one SIP account for termination
or I have to setup individual account for all of
Dial(SIP/904)calls whoever logged on as john.
You could define a variable in extensions.conf.
Nabeel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
I was wondering if there is a way to select the outbound
trunk based on the extension that making the call.
Set the context in the sip.conf file for that user to a context in
extensions.conf that only has entries for dialing out through specific
providers.
Nabeel
Does anyone else have this problem? Is there a workaround?
Yeah, I had this problem when I added a lot of SIP register statements
and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the
problem. It seem * was getting stuck waiting for DNS lookups.
Nabeel
Is there any way I can make a collect call without having a
account with anyone?
1-800-COLLECT.
Nabeel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
Hello.
I have ASTCC running fine on one of my servers. I was trying to install
it on another server but am having issues. I checked out ASTCC from the
CVS and can get to the astcc-admin.cgi. However, none of the
configuration values are being remembered when I hit Save. Where does
the CGI script
I have ASTCC running fine on one of my servers. I was trying
to install it on another server but am having issues. I
checked out ASTCC from the CVS and can get to the
astcc-admin.cgi. However, none of the configuration values
are being remembered when I hit Save. Where does the CGI script
Is there any way to detect if a user has a
mailbox? I want to send all call which match _14XXX to
voicemail except if the user doesn't have a voicemail box...
This is what I have:
_3XXX,1,Dial(SIP/${EXTEN})
_3XXX,2,Voicemail(${EXTEN})
_3XXX,3,Hangup
_3XXX,102,Voicemail(${EXTEN})
Could you expand on your comments, or provide a link / paste
in a sample extensions.conf to show how this would be set up?
I just realized this is not exactly what you want, but let's try it
anyways:
[global]
200 = SIP/john
201 = SIP/mary
[somecontext]
exten = 200,1,Dial(${200})
exten =
Matt Darnell wrote:
Does anyone know of an * plug in that will prompt a user for an
account code when they make a long distance call?
Look at the Authenticate command.
Do you know if the entered string gets printed out with CDR records?
If you had looked into the Authenticate command, you
Mike Dewey wrote:
I am very interested in what you came up with for a 2.5mm
to RJ-10 adapter.
Yes, I built a 2.5mm cellphone headset to Cisco 7960 RJ10 headset jack
adapter. If I follow the numbering system used on
http://www.mml.uni-hannover.de/einhorn/headset/index_e.html, this is how
it
communications and remove nat=yes from the entry for the SIP device inside
NAT2.
To set the SPA to give the correct IP, enable STUN, add a STUN server, and
say Yes to Substitue VIA Addr.
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
In your second option using a STUN server would I need to setup my
own STUN server?
No, use FWD or xten's STUN servers.
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
___
Asterisk-Users mailing
Is there a search engine for this mailing list ?
Google for:
site:lists.digium.com search terms
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
___
Asterisk-Users mailing list
Asterisk-Users
to the timeout
extension which is included in the context.
Regardless, posting more than just the last message from the CLI would be
helpful.
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
___
Asterisk-Users
.
--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990
cisco.tcl
Description: Binary data
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo
).
My question is, do I need ztdummy at all? I don't intend to use
conferencing at all.
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk
to specify alternate TFTP
which I specified as the IP of my PC.
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
do I upgrade firmware?
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo
because the example
config shows a numeric string only.
Finally, how do I upgrade firmware?
I still am not sure how to check and/or upgrade firmware.
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
kind of ATA
for the FXS? Will obviously save a whole bunch of money, but will there
be significant added complexity?
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
___
Asterisk
the Clipcomm CG-410 4-port FXO gateway
(http://www.voipsupply.com/product_info.php?products_id=241)?
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
___
Asterisk-Users mailing list
gateway, set PSTN Caller ID Pattern to *, then set
call-forwarding under PSTN User to:
Cfwd Sel1 Caller: *
Cfwd Sel1 Dest: 123
where 123 is an extension in the context that the SIP account on the *
server is in.
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
that
the SPA-3000 does not pick up the SPA-3000 line until after the
extension/* picks up)?
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users
I have experienced nothing but grief when trying to set up
the PSTN part of the SPA-3000. Everything from crackly audio to fast
busies.
BTW I take that back. I spent an hour on this after posting my last
email, and with a little tweaking, everything seems to be working well
now.
--
Nabeel
experience!
Has anybody tried the Clipcomm CG-410? I am considering getting one but
would like to know if anyone has any good/bad experiences with it?
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
a call?
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
What happens on the phone is that I hear voiptalk.org's
greeting and after they hang up, I hear my own invalid
extensions message.
That's because your _X. extension has a 2nd line. Reduce it to
_X.,1,Playback(invalid) and you won't have that problem.
--
Nabeel Jafferali
tel: 416.491.9136
.
I'm not the original poster but think I've got the same issue.
Why does reducing that to a single line solve the problem?
Because otherwise when you call any number and that call ends, it goes
to _X.,2,... because that matches!
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new
or when the call is over, it would go to
not-in-service,_.,2 - since that is a matched next step in the
dialplan.
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
___
Asterisk-Users
(except 18005551212)
and then, if you are allowing only US/Canada calls, you would want to
block all the other NANPA area codes (Jamaica, Trinidad, etc.)
one-by-one.
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
dtmfmode=inband
canreinvite=no
[bv-in-2]
type=friend
host=147.135.0.128
context=from-bv
dtmfmode=inband
canreinvite=no
[bv-in-3]
type=friend
host=147.135.4.128
context=from-bv
dtmfmode=inband
canreinvite=no
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email
This is not just a problem with LiveVOIP -
for some other countries where VoipJet is primary I've had similar
problems).
Are there any ways to get around this problem? Is there a way to timeout
if ringing doesn't happen in 5 secs (for example) and go to the backup
provider?
--
Nabeel Jafferali
tel
file is wrong.
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk
Anyone in the Toronto area interested in getting together to share
notes and swap war stories?
One of the other guys in Toronto interested in * put together a
meetup.com group. Please join in and we can see where to go from there.
http://opensource.meetup.com/42/
--
Nabeel Jafferali
tel
Are there any ways to get around this problem? Is there a way
to timeout if ringing doesn't happen in 5 secs (for
example) and go to the backup provider?
Anyone?
--
Nabeel Jafferali
tel: 416.491.9136 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
on
the card, it's just blank. Any ideas?
- When does the 2nd, 3rd and 4th trunk get used? I have 1-2
international destinations that don't go through and eventually time
out, but ASTCC just says the party is not answering since the IAX
channel hung up.
--
Nabeel Jafferali
tel: 416.491.9136 (toronto
How do I resolve this once and for all?
I had the same problem on a non-loaded machine running Gentoo. I simply
upgrade mpg123 (using the emerge command in Gentoo) and the problem went
away. You might want to try to reinstall/upgrade mpg123 and see if that
helps.
--
Nabeel Jafferali
tel
a NAT), but I
thought I would let you know anyways.
--
Nabeel Jafferali
tel: 416.628.9342 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
}/${EXTEN:1},,)
exten = _${DIAL_OUT}011.,3,Congestion
exten = _${DIAL_OUT}011.,103,Macro(outisbusy)
I don't see you calling the astcc.agi anywhere.
--
Nabeel Jafferali
tel: 416.628.9342 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
'callstart char(40)'
Thank you very much. That worked perfectly.
--
Nabeel Jafferali
tel: 416.628.9342 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
be a
timeout to confirm the call has been passed on to the PSTN
successfully).
I have submitted a feature request:
http://bugs.digium.com/bug_view_page.php?bug_id=0003282
--
Nabeel Jafferali
tel: 416.628.9342 (toronto)
646.225.7426 (new york)
fwd: 46990
email/msn : nabeelatjafferali.net
does anybody knows from where I can get an list of
international area codes incl. the mobile numbers?
The way I did it is to get the rate tables of one of the IAX providers
(LiveVOIP, VoipJet, NuFone come to mind).
--
Nabeel Jafferali
tel: 416.628.9342 (toronto)
646.225.7426 (new york
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
a call using a Direct Dial IP account in X-Lite,
I get the following error in *:
Failed to authenticate user Nabeel
sip:192.168.78.50;tag=1234567890
How do I tell * to allow any un-authenticated user access to the default
context?
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1
9105551212 = 1234,Gary Carr,[EMAIL PROTECTED],attach=yes
Syntax is:
Mailbox = password,Name,email,pageremail,options
So, that should have been (added delete, it's a good idea if you're
attaching).
9105551212 = 1234,Gary Carr,[EMAIL PROTECTED],,attach=yes|delete=yes
--
Nabeel Jafferali
Tel
Did someone know where to find the SIP image for Packet8 DTA
310 box to work with Asterisk ??
You need 0x to define the SIP settings and 0x1234 to get it to work.
Both are available in the VoIP forums at DSLReports.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225
= _00XX.,1,DeadAGI(astcc.agi,${ACCOUNTCODE},011${EXTEN:2})
Once the call is finished I would like to have the balance
shown in the display by sending a sip message to the phone(if
possible otherwise not important).
This would require adding code to the AGI, if it's even possible.
--
Nabeel
I was looking for the SIP IOS of the Cisco IP Phone but i
can´t find it in the cisco web page.
What is IOS? Am I the only one who uses Cisco phones and doesn't know that
acronym?
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN
- VoIP taxes/regulation in Canada
- * compatible hardware vendors in Canada
then it should be a welcome addition to the * community. However, any
general question should be directed to the -users list.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Just my CAD$0.02 though.
C'mon, at least throw in a loonie :P
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users
to actually use it.
AND STOP USING HTML! :P
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
user: aaabbb
pass: cccddd
register
iax.conf:
=
[623] ; IAXy
iax.conf should read:
[aaabbb]
username=aaabbb
...
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
Does anyone know how many simultaneous calls per proxy I can
recieve/place on a Cisco 7960G?
Two.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
___
Asterisk-Users
buy one of the adapters for the standard headset
connector and then buy the Bluetooth adapter with those connectors?
Any help would be appreciated.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
this is OT, but can you recommend an email program for Windows
that does something like that?
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
___
Asterisk-Users mailing list
and cordless phones)? Any idea what the pinout for that would
be?
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
, but it didn't work. Reordering
the mySQL table so these 8 non-NANPA catch-alls appeared at the top of
the table (before the 1416 and other NANPA entries) fixed it though.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
Did you ever figure a way around this? It would be a good
time to test since LiveVoip is having some issues today.
No, I'm afraid I never found a solution. I posted a feature request on
Mantis but I guess there isn't enough interest.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1
to complete the call due to PSTN issues (as opposed to IP issues)
and is just waiting for progress (i.e. the call hasn't failed).
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
I am trying to get a SPA-3000 to work behind NAT - for the
sake of the exercice.
Post the relevant entries from sip.conf and extensions.conf, and the
relevant fields from the SPA-3000 Line 1 tab.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
under NAT Support Parameters on the SIP tab and put in a STUN server
(stun.xten.net comes to mind). Then, enable NAT Mapping and NAT Keep
Alive on the Line 1 tab. Remove nat=yes from sip.conf and try again.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD
Many thanks Julian.
Are you looking for the pinout for a single plug 2.5mm (cellphone)
headset or a dual plug 3.5mm (computer) headset?
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
pretty sure is the same as a standard telephone headset
jack.
You could try both - that's what I did when building my single plug
2.5mm (cellphone) headset to Cisco 7960 headset adaptor.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN
] with the right host IP.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman
The issue I am having is that the READ command uses # as a
termination symbol. Is there any other way I can accept # as
input or must I use a digit instead?
Use WaitExten() instead to get the extension.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD
that * received the SIP messages from for RTP traffic (use via
IP address and via port).
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users
How can I disable native bridge for IAX calls?
I know for SIP you can put 'canreinvite=no' but this does not work.
notransfer=yes
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
.
--
Nabeel Jafferali
Tel: +1 (416) 628-9342 Toronto
+1 (646) 225-7426 New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
4. Scnet.net has 5 pages website (quite a work for ISP), that
any kid could create in 1h
scnet.net is Server Central, a data centre where my host (HostForWeb),
among others, maintains their servers. I do know it is a reliable data
centre and I doubt is in any way related to iax.cc and/or
Anybody know a good IAX provider for Canadian DIDs?
I currently use Xetricom for Toronto DIDs (C$7.50 each). I also know of
someone who can provide a Toronto DID with unlimited* GTA calling for
C$20.
Nabeel
___
Asterisk-Users mailing list
Does AreskiCC allow the card number to be passed from the dialplan, like
ASTCC does?
Nabeel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
I am experimenting with my * server to use SIP with my long-distance
providers instead of IAX, so that the media path is from the end user
straight to the provider's gateway (hopefully reducing my bandwidth
consumption). I have it working with VoicePulse Connect but SetCIDNum
doesn't appear to
Still trying to get NAT working.
Try adding a canreinvite=no.
Nabeel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
I would think the BudgeTone would be good, but then I've read
so many people complaining about them, and some people seem
to recommend the Sipura adapters.
For agent use, the BudgeTone's lack of three-way calling would be an
issue.
Nabeel
___
Whilst I agree with Joe, has anybody actually been able to
sucessfuly get the Cisco 7940's/7960's to register into *?
Yeah, I've been using a 7960 with * since November.
Nabeel
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I use the latest ASTCC version, but I do not see how you can
use different cards / different prices. Can you explain that for me,
please?
The cards have a field called markup, that would mark up the price
specified in the routes table by a certain percentage. So it's a
card-specific,
[invalid]
exten = _.,1,Answer
exten = _.,2,Playback(pbx-invalid)
exten = _.,3,Hangup()
asterisk is playing invalid message twice, WHY?
-- Executing Answer(SIP/11-df84, ) in new stack
-- Executing Playback(SIP/11-df84, pbx-invalid) in new stack
-- Playing 'pbx-invalid' (language
.)
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeelatjafferali.net
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
and upgrade to (not higher
than) v1234. I haven't tried it with *, but I assume it should work.
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeelatjafferali.net
___
Asterisk-Users mailing list
[EMAIL PROTECTED
phones?
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeelatjafferali.net
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
Tone-100). Is there anything out there
that can do that?
I have not tried it, but this claims to do what you need:
http://www.yottadot.org/download.php?op=viewsdownloadsid=10
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeelatjafferali.net
their
home router has a real IP, not a private IP. So, for some reason, STUN
(or something else) is seeing their IP as their ISP's router. It almost
seems like their connection is double-NATted?
I am unsure what steps to take next, so any help would be greatly
appreciated.
--
Nabeel Jafferali
tel
is plugged into another NAT router -
double NATted? The * server is on a public IP.
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeelatjafferali.net
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
the problem of NAT
mapping.
Besides the IAXy, are there any other IAX adapter devices? Any being
planned?
I realize the IAXy came out a little while ago - are there any obvious
features missing from it? Where's a good place to pick up 4-5 of these
(I'm in Canada)?
--
Nabeel Jafferali
tel
Try this:
canreinvite=no
As I mentioned in my initial email, I tried that, and adding that line
eliminated 1 of 2 problems. The other problem, that of one-way audio
when a call is carried into the server from and IAX gateway provider to
that SIP client, will not go away.
--
Nabeel Jafferali
the * server?
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeelatjafferali.net
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
that.
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeelatjafferali.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
for customers and administrators.
Something that can tie in to one of the existing management GUIs would
be a big plus.
Any ideas?
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeelatjafferali.net
. How do I remove the modules that are in CVS-HEAD?
What's the best way to do clean up everything and do a fresh install of
Asterisk stable?
--
Nabeel Jafferali
tel: 647.722.8457 x201
718.606.4190 x201
fwd: 46990 x201
email/msn: nabeelatjafferali.net
smime.p7s
Description: S/MIME
If you have, in sip.conf, a register = blah:[EMAIL PROTECTED]/12345, you
would also have:
[blah]
host=sip.blah.com
context=from-blah
Then, in extensions.conf, you would have:
[from-blah]
exten = 12345,1,Dial(whatever)
...
Nabeel
From: [EMAIL
1 - 100 of 196 matches
Mail list logo
