Dear all,
In Sip.conf file how to setup incoming calls not to use
authentication?
Please provide some steps to do it..
Thanks...
Regards,
Velusamy
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I have a new Asterisk system going into production next week and I'm a
bit stumped as to the best way to handle the Dialplans for it.
The Asterisk system is replacing 4 separate PSTN lines with both SIP
PSTN inputs. The setting up of the dial plan is giving me some design
headaches, which
On 1 Aug 2009, at 08:32, Myles Wakeham wrote:
I have a new Asterisk system going into production next week and I'm a
bit stumped as to the best way to handle the Dialplans for it.
The Asterisk system is replacing 4 separate PSTN lines with both SIP
PSTN inputs. The setting up of the dial
http://www.voip-info.org/wiki/view/Asterisk+sip+insecure
On Sat, Aug 1, 2009 at 12:52 PM, velusamy velu velu.techni...@gmail.comwrote:
Dear all,
In Sip.conf file how to setup incoming calls not to use
authentication?
Please provide some steps to do it..
Thanks...
Regards,
Hack the new asterisk server - see below:
Interested in joining the friendly global Free SW HW Culture communities in a
global Voice meeting? You´re invited. :)
You can join from your home, or better: get a local meeting together. Tip: a
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Hi,
I remember reading that Asterisk allows only 100 simultaneous calls. Is
that correct?
If it is so, how is it possible to have a conference call with more then
100 users? I think I read here that some people managed to have 500
people in a conf room...
Or, how do I increase this limit? Is it
Emrah wrote:
Doug,
Thanks for the suggestion.
I know there are plenty of workarounds there, I am not asking how to do
it because I know how to do it too.
If it can be easily done with the dial plan, then it's not likely to be
added as a feature.
Doug
--
Ben Franklin quote:
Those
On Saturday 01 August 2009 00:42:05 hadi motamedi wrote:
Can you please let us know how to configure Asterisk to recognize
extensions starting with the hash key ?
There's no need to do anything special. As the hash key does not start
comments in Asterisk configuration files (which sounds like
Hi,
I am looking for H248 support in Asterisk, does anyone know if there is
anything available for that?
Thanks,
Mark
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mark morreny wrote:
I am looking for H248 support in Asterisk, does anyone know if there is
anything available for that?
No.
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Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
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Tilghman Lesher wrote:
On Saturday 01 August 2009 00:42:05 hadi motamedi wrote:
Can you please let us know how to configure Asterisk to recognize
extensions starting with the hash key ?
There's no need to do anything special. As the hash key does not start
comments in Asterisk
On 1 Aug 2009, at 08:32, Myles Wakeham wrote:
I have a new Asterisk system going into production next week and I'm a
bit stumped as to the best way to handle the Dialplans for it.
I have separate entire phone 'systems' for each incoming DID. For
example, this one system will handle 4
Tim wrote:
Sure, have a top level context that inbound calls from the ITSP go into:
[from-ITSP]
exten = 2125551112,1,Goto(companya,${EXTEN},1)
exten = 212555,1,Goto(companyb,${EXTEN},1)
; then separate contexts for each company:
[companya]
extern = 212555,1,.
extern =
The calls with all come in to one context, in this example from-ITSP. From
there you send it to one context per company. As the example shows. I might
do:
; All calls from ITSP come to here and you route each number somewhere else
[from-ITSP]
exten = 2125551112,1,Goto(companya,menu,1)
exten =
This question has been asked thousands of time on this list. You mat want
to search the archive, but to sum it all, there is no limit as far as calls
on an Asterisk server. It all depends on your server's specs, and how it is
setup. A celeron processor with 256Mb Ram could handle a fews calls
Hello!
I am wondering how to configure Asterisk and devices so I can use
different codecs for upstream and downstream packets.
Thank you,
Elliot
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Hi,
I am using SNOM phones with Asterisks for few years. They used go occasionaly
NR, and I would reregister them from the phone web interface. But it started
doing frequently for last few months.
Here are SNOM Phone and the firmware version;
snom190-SIP - Version-Code: snom190-SIP 3.56m
Elliot Murdock wrote:
I am wondering how to configure Asterisk and devices so I can use
different codecs for upstream and downstream packets.
You can't. With SIP as the channel technology, at least, the SDP
negotiation model demands a uniform codec from both sides.
I guess there's nothing to
Pascal Bruno wrote:
Also you can cluster Asterisk servers making its capacity
pretty much unlimited, if you know what you doing nd if you doing it right.
I think that's really the key. Don't do anything that's particularly
hostwise, i.e. is limited to the scope of one server. Standard
Hello,
Thank you...do you know if IAX can do this?
The reason for doing is this is to get over the adsl upload/download
discrepancy. While G711 gives terrific quality, it is not always that
feasible for the upload direction, which has much more limited
bandwidth. Accordingly, it would be
Elliot Murdock wrote:
Thank you...do you know if IAX can do this?
The reason for doing is this is to get over the adsl upload/download
discrepancy. While G711 gives terrific quality, it is not always that
feasible for the upload direction, which has much more limited
bandwidth.
---
Hi,
I got hit with a funny one today; my configuration, which had been
running fine for many weeks, suddenly stopped registering with
Broadvoice's SIP. I switched among BV SIP hosts, verified my account
was OK, even tried upgrading Asterisk. The weird thing was that I could
FWIW, my broadvoice setup ( and I just upgraded to 1.4.26 to play with Skype
channels and verified that Broadvoice still works )
register=781zzzn...@sip.broadvoice.com:p
assword:781zzzn...@sip.broadvoice.com/781zzz
[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
Nice job. It worked right away for me with my 10 channel trial license.
Asterisk 1.4.26
I'm already building a dtmf access menu to bridge to my SIP world :-)
As much I hate Skype for being a closed system, it would make the ultimate
remote Asterisk extension as Skype drills through so many
On Sat, Aug 1, 2009 at 12:04 AM, hadi motamedimotamed...@gmail.com wrote:
Dear David
I appreciate your reply . Pleae find attached our current extensions.conf
file . Can you please do me favor and let me know where I am expected to put
your proposed line for defining the timeout param ?
oi geli wrote:
Hi,
I am using SNOM phones with Asterisks for few years. They used go occasionaly
NR, and I would reregister them from the phone web interface. But it started
doing frequently for last few months.
Here are SNOM Phone and the firmware version;
snom190-SIP - Version-Code:
Unfortunately for me, I cannot register my license. Kept saying:
Could not generate Host-ID.
Make sure that you have eth0 enabled.
Any help would be appreciated
On Sat, Aug 1, 2009 at 9:01 PM, Tom Browning ttbrown...@gmail.com wrote:
Nice job. It worked right away for me with my 10
Unfortunately for me, I cannot register my license. Kept saying:
Could not generate Host-ID.
Make sure that you have eth0 enabled.
On Sat, Aug 1, 2009 at 9:01 PM, Tom Browning ttbrown...@gmail.com wrote:
Nice job. It worked right away for me with my 10 channel trial license.
Asterisk 1.4.26
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