Displaying 20 results from an estimated 800 matches similar to: "SIP, X-Lite"
2005 Sep 14
2
Starting From Scratch
Hello all:
For fun, I am learning about Asterisk, and trying to get Asterisk
working at my house. I installed Asterisk@Home. It seems to be
functioning fine. I installed a couple of softphones, and have them
registered with Asterisk. I actually work for a CLEC, and I have
registered my Asterisk box with SER (which I don't begin to understand
yet) at the office. In order to try to
2003 Mar 14
3
SIP registrations
Can asterisk act as a SIP registrar or location server?
I would like to be able for a user agent(client) to register with
whatever client they are using as "username@domain-name.com". Rather
than the entry/username/password that is setup in the sip.conf file.
That way a user could log into any SIP enable client and their calls
would follow them around.
I have read the sip.conf man pages
2004 Nov 26
1
direct asterisk to asterisk SIP calls without external SIP provider
Hi all,
I have a small system of two hardware boxes (residential gateways)
running Linux with Asterisk on them. Each RG has some FXS ports to which
analog telephones can be connected.
I already had a working system including an external SIP provider, where
both RGs would register to that provider with a telephone number and
they could call each other via that telephone number. Each RG had a line
2004 Nov 22
0
How to configure the Asterisk server such that a FXS phone can talk to SIP client?
Hi,
Could you please help me!! I am trying to configure the Asterisk server.
I have a analog phone connected to a FXS port of a Cisco 3745 router. This router is connected to a Asterisk server via Fast Ethernet interface. I am trying to make a call from the analog phone to a SIP client. This SIP client is registered to the Asterisk server.
Analog phone number: 999
SIP client : 202
Sip client IP
2004 Nov 27
0
Failed to WWW-authenticate on INVITE
I'm having trouble connecting a asterisk server to a SIP Express router.
Inbound calls to my asterisk server works just fine, but when i try to
make outbound calls I get the following error message:
Nov 27 22:40:48 NOTICE[4687]: chan_sip2.c:7967 handle_response: Failed to
WWW-authenticate on INVITE to '"username"
<sip:username@mysipprovider>;tag=as5399a078'
I'm
2003 Mar 09
6
DTMF detection on SIP provider ?
Hi..
I just wondering why DTMF are not recognized by aterisk on incoming calls
from my SIP provider ...
ANy suggesteions ?`
/Mike
2006 Apr 19
1
Fwd: sip.conf and jump from register to the extension
Hi,
the documentation of sip.conf is telling me this:
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
In reality it jumps to the extension 1234 in the context and not to s
So it is much more complicate to write an proper dialplan.
Is this an bug or is the documentation not up to date?
best regards
Thomas
2015 Jan 25
1
tinc Digest, Vol 123, Issue 13
Lastest Modifications:
HOST A:
Removed 2 route add as you suggested, and it's still working.
HOST B:
This host has a openwrt as gateway and I added few days ago as you
suggested on freenode and this is HOST B Gateway route
Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use
Iface
0.0.0.0 192.168.1.1 0.0.0.0 UG 10 0
2004 Apr 23
6
Polycom registration
I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone
registered on an asterisk box but am having no luck. I get the
following errors 192.168.22.196 being the phone and 22.254 being the
asterisk box..
Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request:
Registration from '"110" <sip:192.168.22.196@192.168.22.254>' failed for
2003 Aug 26
0
TDM10M && Siemens Euroset 2015
Hi all,
--------
I have installed a TDM400 with one active FXS port (TDM10B) an connected
it to a Siemens Euroset 2015 analogue phone.
I have installed some smom IP phones to the network as well and
configured them as usual (sip.conf). For configuring the TDM10B I have
used FXO signalling in /etc/zaptel.conf and in
/etc/asterisk/zapata.conf. I definded the TDM channel and the Snom
phones to the
2003 Nov 24
0
SIP channel modification
If you update your source from the CVS, you'll get a new SIP channel
that supports a new syntax for SIP calls in extensions.conf
If you define a SIP peer in sip conf, like
[mysipprovider]
...
You can now use
dial(SIP/mysipprovider/extension)
Where the part "mysipprovider" is related to the sip.conf section.
Also, you can dial any SIP URL by
2008 Sep 18
2
o2hb_do_disk_heartbeat:982:ERROR
Hi everyone;
I have a problem on my 10 nodes cluster with ocfs2 1.2.9 and the OS is RHEL 4.7 AS.
9 nodes can start o2cb service and mount san disks on startup however one node can not do that. My cluster configuration is :
node:
ip_port = 7777
ip_address = 192.168.5.1
number = 0
name = fa01
cluster = ocfs2
node:
ip_port =
2004 Jun 16
5
Failed to authenticate on INVITE
Hi,
I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04).
These two boxes talk to eachother via sip, not iax. Since the upgrade, I
get the error "Failed to authenticate on INVITE" trying to make calls to/from
either box. Removing the secret from each box's sip config seems to work but
is utterly braindead.
Has anyone seen this?
- Eric
2002 Apr 25
1
Routing between two tunnels
Hi!
Me and two friends are trying to get a VPN working, but we cant get routing
between two tunnels.
This is how it looks, all servers (192.168.*.1) are running IP Masquerade to
enable the other computers behind them to access the internet.
Both elayne and glenn are connecting to melc, and the tunnel between melc
and glenn are running TCPOnly because that glenn doesnt have a public IP
(it's
2006 Mar 29
5
Asterisk Between PBX and FXS
Hi guys,
I''m setting up asterisk to run with another pbx server. This pbx server
support a feature that allows 2 extensions connect to the same FXS. No I put
asterisk in the middle.
Asterisk receives the call and dial to a SIP/peer.
How the pbx installed support 2 extensions to one fxs... How can I figure out
in asterisk which extension was dialed before the call came to asterisk?
2003 Oct 27
4
Groups in *
Hi list!
I have a little question about groups and Asterisk ... is there anyone out there that can say if Asterisk can do any of this;
We have a customer that want call handling we cant give him with a traditional PBX, and I'm running Asterisk @home so I thought I could give it a try ...
The customer wants that incoming call should go to one group with some phones in it, if the group is
2010 Dec 14
1
Asterisk + VOSP account working configuration?
Hello
I'm having a difficult time finding precisely what to put in
sip.conf and extensions.conf (and possibly other files) to get a
working configuration to connect an Asterisk (1.4) server to a VoIP
provider with the Asterisk server + SIP clients located in a private
LAN behind a NAT router:
http://img560.imageshack.us/img560/3749/asterisknat.png
Would someone have a full, direct (ie.
2004 Jul 18
4
Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi All
Total noob on the list so all help appreciated....
I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows).
I've plugged in two Cisco 7960 phones....
The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......
2006 Dec 18
0
pap2/wrt54gs/asterisk
I am having trouble setting this system up and wonder if some one help me.
Does anyone know what is missing if anything to get 2 phones on my
asterisk home server to be able to call each other.
I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2
extensions 5060/5061, this is on the lan side of my gateway/router
WRT54G 192.168.1.1
BusyBox v1.00 (2006.11.07-01:40+0000)
2007 Sep 28
0
[LLVMdev] Crash on accessing deleted MBBs (new backend)
Replying to my self here. It seems I was missing an isTerminator = 1
on the branch instruction in question, so LLVM didn't know that the
instruction terminated a basic block. Does that make sense, or is just
masking some other problem?
Thanks,
Andreas
On 9/28/07, Andreas Fredriksson <deplinenoise at gmail.com> wrote:
> Hi,
> I'm trying to write up my little m68k backend