Well, unfortunately i did not dig much into why/how it worked with
openvpn, but it did work for me with default setup.I think you may need to
set constant ports instead of random ports.
Thanks,
Vivek
On 11/9/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi Friends;
Actually I would appreciate
Hi Friends;
Actually I would appreciate if Vivek can advise if the
VPN resolved the RTP packets in the SIP Trunk between
Asterisk and another softswitch? In other words,
openvpn helpful in NAT cases in what exactly? As
without VPN, I was able to establish a call but
without voice or with complete
On Nov 9, 2007 2:53 PM, bilal ghayyad wrote:
[...]
From the other side, I think that baji is talking about
something else than the IP Trunk, he is talking
about outbound [...]
correct, I was responding to Gabriel's post on being
registered w/ SIP provider accepting inbound, but
having
after a copious loss of follicles :-), I finally got outbound working.
Basically the channel statement in the call file needs to have the
number to be called. For eg., in test.call format the statement
as follows :
Channel: SIP/3012345678@your-sip-provider
And there is no need for a
yeah i found openvpn helpful in NAT cases.
-Vivek
On 11/6/07, Baji Panchumarti [EMAIL PROTECTED] wrote:
after a copious loss of follicles :-), I finally got outbound working.
Basically the channel statement in the call file needs to have the
number to be called. For eg., in test.call
: Monday, October 29, 2007 6:54 PM
Subject: Re: [asterisk-users] Everyone is
busy/congested: IP Trunk
No:
register = abc:[EMAIL PROTECTED]
[peer]
host=zzz
Its possible to make mistakes and typos you know.
Maybe you can post
your config file and we can help you.
On 10/26/07, bilal ghayyad
- Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, October 29, 2007 6:54 PM
Subject: Re: [asterisk-users] Everyone is busy/congested: IP Trunk
No:
register = abc:[EMAIL PROTECTED]
[peer]
host=zzz
Its possible to make mistakes and typos you know. Maybe you can post
No:
register = abc:[EMAIL PROTECTED]
[peer]
host=zzz
Its possible to make mistakes and typos you know. Maybe you can post
your config file and we can help you.
On 10/26/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi Pablo;
How the IP address will be wrong, and asterisk able to
do
On 10/27/07, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi Pablo;
How the IP address will be wrong, and asterisk able to
do registeration on the destination?
If the IP address wrong, so I will not be able to
register on that IP address.
Hi
i see 2 causes
1. it could be Dialplan issue (
Hi List;
I established an SIP IP Trunk between Asterisk and
another softswitch (asterisk registered on the
softswitch successfully) and I saw this on the
softswitch.
From firefly softphone, I was need to do a call to be
via this softswitch (ofcourse, the softphone will send
for asterisk and
On Fri, Oct 26, 2007 at 06:55:12AM -0700, bilal ghayyad wrote:
Hi List;
Ip address to destination?
Unable to create channel of type SIP (cause 3 - No
route to destination)
i think you have the wrong ip information
I established an SIP IP Trunk between Asterisk and
another softswitch
Hi Pablo;
How the IP address will be wrong, and asterisk able to
do registeration on the destination?
If the IP address wrong, so I will not be able to
register on that IP address.
Regards
Bilal
Hi List;
Ip address to destination?
Unable to create channel of type SIP (cause 3 - No
route
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