Displaying 20 results from an estimated 5000 matches similar to: "SIP IP Trunk, between Asterisk and Softswitch"
2009 Jul 05
1
SIP IP-Trunk to be authenticated based on username and password, not IP address
Hi List;
How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed
2007 Oct 23
2
register => to let Asterisk register to another softswitch via SIP
Dear Alex;
Thanks alot for your nice help.
This is if I need to let Asterisk register with
another softswitch (so I used register =>), what if I
need asterisk to send call for the softswitch without
register to it (directly)? If I removed the register
=> then how it will distiguish the IP address in the
"host" at the [sip_trunk] is the IP address of the
softswitch that need to
2007 Jul 03
5
Determining the used codec for the IP Trunk (SIP Trunk)
Hi List;
Where I determine the codec to be used for the SIP
Trunk (between Asterik and another SIP softswitch)?
Regards
Bilal
____________________________________________________________________________________
Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out.
http://answers.yahoo.com/dir/?link=list&sid=396545433
2007 Jun 28
2
Linking Asterisk with another SIP PBX (or SIP Softswitch)
Hi List;
If I need to do a trunk between Asterisk and another
SIP softswitch (so Asterisk will send a SIP calls to
that softswitch), then I have to configure this on the
sip.conf file or where exactly? And is it the same
when I configure iax trunk?
Should I determine the context in this case for this
SIP trunk?
Regards
Bilal
2007 Sep 09
3
nat=yes
Hi List;
If I set nat=yes, then asterisk will send the packets
to the public IP address or to the private IP address
(which will be for the endpoint that is behind the
nating)?
And by setting the nat=yes, then what exactly will be
ignored at asterisk side when reading the
registeration messages from the endpoint?
Any help.
Regards
Bilal
2007 Aug 26
1
Is it possible to register without sending the password?
Hi List;
I noticed that if I disabled secret in the context by
placing ( ; ) before it, then at the asterisk the log
will be:
-- Registered SIP 'bilal_sip' at 0.0.0.0 port 5060
expired
The IP address of the endpoint was not captured!!!
Why?
If secret enabled, then some endpoints can not
register (maybe due to compatibility in reading the
negotiation packets), so what is the solution?
2007 Oct 19
1
Using register => to let Asterisk register to another softswitch via SIP
Hi All;
Alot of softswitches or PBX's does not accept to
manipulate any SIP call without being registered
firstly. So that means, I need asterisk to register
firstly then I can route my calls to that SIP trunk.
In IAX2, we use the register => , so what shall we do
in Asterisk? And how its format will be (if we will
use register)? Or what is the solution?
Regards
Bilal
2007 Jul 12
0
No subject
help me in another issue related also to registering
asterisk with another softswitch:
A) If nat=yes, then I have to set canreinvite=no to be
able to register, correct?
B) In case of using firefly softphone, how it possible
to set it to have nat=yes (at the firefly it self and
not at the sip user configuration section)? As most of
the sip endpoint give an option to set nat=yes and so
on, how it
2007 Aug 27
2
Is it possible to register without sending the password
Dear Philipp;
Kindly find the part of the configuration as below:
[general]
allow=all
disallow is comment by ( ; ).
[bilal_sip]
type=friend
context=internal
host=dynamic
canreinvite=no
dtmfmode=rfc2833
So where is the problem? The endpoint does not
register and nothing appear on trace level 3. And the
amazing thing that if the endpoint send wrong username
(for example: bilal_sip100) then it
2007 Jul 24
10
What is the best softphone work with Asterisk
Hi List;
I need to configure a softphone to be client and use
it with Asterisk, which is the recommended one? Is it
iax2?
Regards
Bilal
____________________________________________________________________________________
Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.
2007 Aug 20
2
Firefly IAX2 configuration
Hi List;
I am using Firefly softphone Version 1.9.9 Build 4521
and I select IAX protocol and did the configuration in
Network1 (and I checked the Active checkbox) as
following:
Server: 192.168.8.4
username: iax2user1
password: password
In the Asterisk, I did the following configuration on
the /etc/asterisk/iax.conf:
[iax2user1]
type=friend
context=internal
username=iax2user1
secret=password
2007 Aug 25
0
SIP endpoint registeration problem
Hi List;
I have a problem when trying to let an SIP ATA
endpoint (got it from broadtel company), I am getting
the following message:
- Registered SIP 'bilal_sip" at 0.0.0.0 port 5060
expires 60
I do not know why it takes it 0.0.0.0 while it has an
IP address (192.168.8.3).
In the sip.conf, the following configuration to the
bilal_sip done:
[bilal_sip]
type=friend
context=internal
2007 Jun 14
3
My Kernel
Hi List;
I did yum install kernel and yum install kernel-devel,
now when I type 'uname' -a I have the following:
[root@localhost /]# 'uname' -a
Linux localhost.localdomain 2.6.15-1.2054_FC5smp #1
SMP Tue Mar 14 16:05:46 EST 2006 i686 i686 i386
GNU/Linux
And when I type rpm -q kernel, then I have the
followig:
[root@localhost /]# rpm - q kernel
kernel-2.6.20-1.2319.fc5
So the
2007 Sep 19
2
what is softswitch
Dear all
what is softswitch what is difference between asterisk and softswitch ??
regards
satish patel
---------------------------------
Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more!
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2007 Dec 02
2
Softswitch digim
Hello averybody,
I'm looking the softswitch in digium website, anyone test the softswitch?
Best Regards
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071202/2440f782/attachment.htm
2007 Jul 27
4
CONSOLE=Phone/phone0 and CONSOLE=Console/dsp and Zap/g2
Hi List;
What the following mean:
CONSOLE=Phone/phone0
CONSOLE=Console/dsp
TRUNK=Zap/g2
I know SIP/John and Zap/1 but I do not know above (I
do not know also the difference between Zap/2 and
Zap/g2)?
Any advise?
Regards
Bilal
____________________________________________________________________________________
Got a little couch potato?
Check out fun summer activities for kids.
2003 Aug 06
2
Semi-newbie question "Softswitch" and Asterisk - Is there a difference?
I've been working in the VoIP industry for just a bit over a year now...
Mostly taking care of the underlying systems. I've now reached the
point where I'm being drawn more and more into the call processing side
of things. My background is in computer and "classic" telephony systems
(DMS250/MTX, DSC 400, T1, channel banks. telabs analog echo supressor
modules and
2003 Nov 07
2
Softswitch
Pardon my ignorance, but I was hoping someone could clear up something for
me.
- For a few POTS lines, digium has a single port card for that, or a T1 card
to a channel bank.
- For 10 or more lines, digium has a T1 or E1 card for that too based on PRI
channels
- For 100's to 1000's of lines, I suspect a soft-switch is in order???
A traditional phone company will sell:
- POTS lines for
2007 Jul 13
1
Can Asterisk hear on two IP addresses? And can I do routing for calls from private to public or public to private IP addresses
Hi List;
Can asterisk hear (receive) calls on two IP addresses?
How?
If yes, then:
If I have a VPN router, and my Asterisk server
connected to two network cards, one has a private IP
address (192.168.0.2) connected to the VPN router
(192.168.0.1) and another network card has a private
IP address (193.111.196.249) connected directly to the
outside default gateway (193.111.196.240), where the
VPN
2003 Oct 24
8
SS7 signaling/Softswitch
I'm confused a bit about the following and was hoping to get some answers on
this group - What is exactly implied when we say asterisk can connect to a PSTN.
Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume
asterisk does not need to do any SS7 signaling and all it does (playing the role
of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct
statement?