On Thu, Apr 21, 2011 at 5:42 AM, Alexandru Oniciuc
alexandru.onic...@trivenet.it wrote:
Dear * users,
in your opinion, when using a * as a public server, is good practice
enabling nat=yes in sip.conf for all the peers?
Can anyone imagine a scenario when enabling this parameter (even for
Hi,
When NAT = YES, Asterisk server will extract IP from the network layer.
When Nat = No, the Asterisk server will respond to the IP in the SIP header. Am
I right?
May be such type of options can be helpful for SIP application developers.
Can't think of a scenario but If it is set to be
On Sun, Apr 24, 2011 at 5:55 AM, Muhammad Ali ali_...@yahoo.com wrote:
Hi,
When NAT = YES, Asterisk server will extract IP from the network layer.
When Nat = No, the Asterisk server will respond to the IP in the SIP
header. Am I right?
May be such type of options can be helpful for SIP
Hi,
I am unsure of what you are saying.
Just for discussion, if one has a control on the insertion of the IP address
in the SIP header, then nat options working can be verified observed.
In the OSI reference model, the Network is layer 3, IP.
Call it Network, layer 3, or IP, it is the same.
Hello,
Thanks for replying.
Answers below:
On 23 April 2011 18:29, Sherwood McGowan sherwood.mcgo...@gmail.com wrote:
On Sat, Apr 23, 2011 at 11:20 AM, Tiago Geada tiago.ge...@gmail.comwrote:
Hi.
Im having trouble setting variables in channel dialplan and re-using them
in Extension
I did more testing.
Here is a portion of extensions.conf on asterisk-pri:
exten = 5,1,Dial(DAHDI/g1/14186939930,30)
exten = 6,1,Answer
exten = 6,2,Wait(30)
exten = 7,1,Dial(DAHDI/g1/14186939930,30,D(132412983#))
Here is an expert from asterisk :
exten = 22,1,Dial(SIP/6@pri,30,D(132412983#))
Hi List,
I am looking to play around with chan_datacard. Any advice on the best
device to test with (that I can find on eBay) ?
Regards,
Dovid
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Hello,
I traced the SIP packets and saw that the only difference was that the
DAHDI channel returns 183 Session progress ( besides the obvious
differences such as the To and from tags in sip , session id and rtp
ports in the SDP ).
I updated my dialplan on asterisk-pri as follows :
exten =
Hello,
I will summarize the current situation. I have reduced the bug to two
asterisk machines. One of which has a PRI card ( DAHDI channels).
The first server calls the second server with SIP. The second server
bridges the SIP channel to a DAHDI channel. When I send DTMFs from the
first
In the following example
exten = _1NXXNXX,1,Set(GROUP(outbound)=myprovider)
exten = _1NXXNXX,n,Set(COUNT=${GROUP_COUNT(myprovider@outbound)})
exten = _1NXXNXX,n,NoOp(There are ${COUNT} calls for myprovider)
exten = _1NXXNXX,n,GotoIf($[ ${COUNT} 2 ]?denied : continue)
exten =
On Sunday 17 April 2011, Stefan Gofferje wrote:
Hi,
has anyone managed to establish an XMPP connection to the facebook
Jabber servers?
I'd like to send messages on missed calls vie FB.
-S
--
(o_ Stefan Gofferje| SCLT, MCP, CCSA
//\ Reg'd Linux User #247167 | VCP
On Sunday 17 April 2011, Stefan Gofferje wrote:
Hi!
Are hints not yet implemented in res_jabber?
I have this here:
exten = 3000,hint,gtalk/gtalk_account/mari....@gmail.com
But the hint doesn't show any difference. It always shows online on the
phone and core show hints always shows
HI,
I am trying to setup a Class 4 termination setup using a kind of channel
hunting scenerio. I have some SIP DID numbers assigned from the local telecom
provider for termination. MY call comes from my wholesale client and lands on a
switch, then it is routed to asterisk. I want asterisk to
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