Displaying 20 results from an estimated 7000 matches similar to: "host=dynamic and defaultip=xxx"
2003 Jun 27
1
defaultip= in sip.conf doesnt work?
I have several (various brand) sip devices with static IP's.
I understand that asterisk will not accept a registration from these
devices if the host= parameter is not set to 'dynamic' in sip.conf.
I want calls to these extensions to be routable even before the device
registers. I understand that is what defaultip= is supposed to do, but
it doesn't work. I get a busy tone when
2004 Jun 24
4
host=dynamic vs host=xxx.xxx.xxx.xxx
Hi all,
This is probably a really stupid question so I apologise in advance; I've
been looking at this all day and after 12hours I've got nowhere fast.
The situation is:
I've got a couple of 7960's on wireless adapters, the wireless network can
be, shall we say, a little flakey. The phones that are wired into the
network directly via ethernet are always registered and work
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
2019 Oct 08
2
defaultexpiry & maxexpiry on peer level
Hello
is it possible to determine the SIP.conf parameters 'defaultexpirty' and
'maxexpiry' on a peer basis ?
My default value is 300 seconds, but some specific SIP-clients can only
send a SIP REGISTER every 3600 seconds. In current configuration these
SIP peers now become "Unreachable" after 300 seconds.
Or is there another way to differentiate ?
Kind regards.
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 Jul 24
2
SIP IP Trunk, between Asterisk and Softswitch
Dear List;
I am trying to create a link between Asterisk and My
softswitch, the link to be SIP Trunk.
I did the below configuration and I do not know if any
one can help me and advise me to have better
configuration to be sure that link is fine. But I do
not know how to determine the SIP username to be sent
for my softswitch as sometimes the softswitch need to
check it.
Also, does asterisk
2006 May 03
2
SIP Phones behind dynamic IPs
Greetings list,
I'm coming across an issue with some of the GXP-2000 phones we have out in
the wild at clients' employees' homes. In most cases they're behind consumer
ADSL NAT routers on a dynamic IP from their ISP.
In a nutshell, the phone is unable to be called unless it's restarted first,
after which it's fine for a good few hours, then it stops working until
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
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
2004 Apr 05
5
Stable Relase Broken ?
All,
I upgraded to the [*] stable release branch.
When I call into the box (confirmed on 2 installations) the
caller no longer hears the ringing. The CLI confirms that
extensions are being 'rung'.
Whassup?
Willy
Willy Wouters
ypOne Publishing
2011 Sep 14
1
Sip re-register / delay problem.
Hello,
For the moment I have the following settings in my sip.conf. I want to
optimize them to archive the following things:
- for the moment all my users will re-register too often. I want that only
lagged users to re-register quickly.
- check from time to time all users but no too often to see if is logged and
can be called.
Overall i want only lagged users to reregister and users with good
2006 Oct 06
3
regexten & regcontext broken for SIP?
Hi ho,
is there anyone out here that is making use of the regcontext and
regexten settings in sip.conf? I've tried this on two Asterisk boxes
(1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1
being created upon SIP client registration, "show dialplan xxx" reveals
no change.
And yes, I have also read and checked bug 7144; if I go down that route
and no
2006 Jan 20
2
Total number of listeners
Hello!
I am using icecast 2.3-kh2 (for the 302 redirect feature), and all my
relays are authenticating with user/pass. Is it possible to know how
many listeners I have for any given mountpoint counting all the relays?
[]s
Pablo
--
Pablo Lorenzzoni <pablo@propus.com.br>
GnuPG ID: 0x268A084D at pgp.mit.edu
http://www.propus.com.br/ - Propus Inform?tica
2015 Aug 05
2
Asterisk uses "Anonymous", but why?
Hi All
I am trying to dial out using SIP and Vonage using the instructions :
<a href="http://www.voip-info.org/wiki/view/Asterisk+and+Vonage" target="_blank"
2010 Aug 02
5
mapping of disconnect reasons
Hi All,
Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 ?Payment Required? from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. For me this is a big issue because my dial plan will look for alternative termination in the event of network error (e.g. reason 3 or 21 which is
2003 Aug 29
6
Asterisk and Cisco 7960
I'm trying to get my Cisco 7960 configured to work with Asterisk, with no
luck. I'm sure I'm missing something very easy... since I know others have
this working. I've stepped through Andy Powell's excellent "Getting Started
with Asterisk", and it works for my X-Lite softphone. My sip.conf entry for
the cisco looks like this:
[cisco]
type=friend
username=cisco
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
2005 Jul 05
4
Uniden UIP 200 and Asterisk.
Hi, I'm new to asterisk, and have a uniden UIP 200 that I got off of ebay.
I'm having trouble getting the phone to register with asterisk. I've tried
a few different settings. I'd be extremely grateful if someone with a
similar setting could give me the sip.conf block for the UIP and the
settings you're using in uniden.txt.
Here's what I have currently:
IP of phone
2007 Jul 26
7
Queue stats
Greetings, list!
My boss would like some statistics on how many calls are answered out of
specific queues during a given time period, and I'm not sure how exactly
to obtain those stats. Here's how our queue system works.
1) Call comes in and enters our 'ring' queue where the phones ring for
15 seconds (caller hears the standard ring tone).
2) After 15 seconds, the caller
2004 May 14
3
SoftPhone to SoftPhone with No Voice
Hello
I Installed Asterisk on RedHat 9. I am currently try to configure minimum with
two softphone talking to each other over the LAN. I am using X-Lite softphones
from xten.com site. I defined 3 phones in sip.conf and also specifies in
extensions.conf file. I am able to ring other computer but there is no voice
exchange ( i can't hear any think except ring). Here is the portion of sip.conf