Displaying 20 results from an estimated 40000 matches similar to: "simple sip question (I think)"
2004 Jul 13
1
SIP authentication bug with insecure= lines?
[wrapping disabled to allow for easier review]
Yet another SIP authentication problem.
I have SER running, and passing calls to a PRI-enabled Asterisk server from a large range of Media Terminal Adapters, and a few other Asterisk systems set up as "clients". I have this PRI-enabled Asterisk server functioning as a very simple media gateway to hand off my toll-free calls to a PRI -
2010 Oct 28
0
Intermittent failure when placing calls - unable to create channel of type SIP
Hello community,
I've been running Asterisk on an embedded device for about six months, and
my operation has been largely trouble-free. I'm hoping I could get some help
with a minor problem:
Every week or three, my PBX gets stuck in a state where it can receive
calls, but it becomes completely unable to originate outgoing calls until I
do a "sip reload". After doing the SIP
2005 Aug 26
1
realtime sip channel configuration -> insecure option
Hi all
I'm trying to figure out what values are valid for the "insecure" option in a
realtime configuration table. The table field is 4 chars long and the actual
valid values for this is longer. Can I modify the field length or has this
changed? Below is where I looked, if I'm not looking in the right place
please let me know.
the field on the table is:
...
`insecure`
2008 Mar 25
1
Sip exten matching based on contact: sip header?
Asterisk: 1.4.17 with sip realtime
Openser 1.3.x
Hi,
I had this setup working fine until I try putting OpenSER in the picture as
a proxy.
Unauthenticated calls go to a PRI based app via a ZAP channel, calls to sip
users get send to them etc. Now with a proxy in the picture asterisk asks
the incoming calls for authentication "407 Proxy Authentication Required".
It seems that the
2009 Jul 28
2
Possibly I don't understand sip peers
I have a carrier who tells me he will be sending me traffic from a wide
range of IP addresses.
so I set up a realtime peer as follows:
[peer]
defaultip=xxx.xxx.xxx.xxx
host=xxx.xxx.xxx.xxx
deny=0.0.0.0/0.0.0.0
allow=xxx.xxx.xxx.0/255.255.255.0
insecure=port,invite
Yes, he's really claiming to originate from any of the IP in the block
When I leave the host blank, we reject calls with a
2010 Feb 17
3
sip.conf - sort order, does it matter
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two identical (sip.conf and extension.conf) files on the same asterisk server and with one set
insecure=invite is working correctly.
When I load the second set of dial plan (sip.conf and
2020 Aug 29
1
401 Unauthorized when originating SIP user exists on remote server
Hi list!
I'm trying to make a SIP test call from Bria and/or 3CXPhone from a PC
behind NAT.
From Bria/3CXPhone I connect to an Asterisk 11.25.0 server on the
internet at 100.100.94.210 with a SIP account "3333" created in sip.conf:
[3333]
type=friend
secret=something
host=dynamic
nat=yes
qualify=no
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
context=voipin
I dial +1234
2012 Aug 02
1
Originate call from cli does not work for SIP line...
I have a SIP line that is working fine when I make calls from IP
phones. I can send and receive calls. The problem is that if I try to
dial from the CLI using the originate command or use an AMI connection
to originate a call I get the following error:
originate SIP/protel-out/0445540881644 application playback tt-monkeys
WARNING[12950]: chan_sip.c:20437 handle_response_invite: Received
2005 May 10
0
outbound PSTN numbers over SIP failing
Hi,
I am currently trying out the asterisk@home (version 1) release of
Asterisk, and I want to configure it as follows:
Calls from regular telephony network (PSTN) come in through my VoIP
provider over SIP and outgoing calls to the PSTN should be routed
through the ViOP provider onto the PSTN network. I thus have no direct
PSTN connection, but only a SIP connection.
Incomming calls
2011 Feb 02
0
SIP Originate on 1.8.X
I am having a problem trying to use originate from the CLI on Asterisk
1.8.2.3. The SIP peer is defined correctly and it works if I dial using
my IP phone. When I try to dial from the CLI I get this message:
pbxoficina*CLI> originate SIP/protel-out/0445540881644 application playback tt-monkeys
[Jan 18 12:00:09] WARNING[3336]: chan_sip.c:19048
handle_response_invite: Received response:
2014 Nov 10
0
Asterisk 11.14.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.14.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.14.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs
2014 Nov 10
0
Asterisk 11.14.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.14.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.14.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs
2007 May 16
1
SIP INVITE failing and AgentCallBackLogin()
Hi List,
Ive got a few * boxes connecting together, one box is doing
AgentCallBackLogin() and then the 2nd box is holding some phones at a remote
site. I have users login to the main box and * shows the user is logged into
a extension that resides on the other box, problem is, when I go to make a
call to a agent, I get
"May 16 05:59:08 NOTICE[13897]: chan_sip.c:9750 handle_response_invite:
2006 Nov 07
3
Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue
Hi All,
I have a lab setup with two asterisk servers and a MAX TNT in the
middle like this:
asterisk sip >< sip TNT pri >< pri asterisk
The TNT is running 11.0.6 and the asterisk servers are running
1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to
asterisk but not the other way. The call from asterisk to pri to tnt
is good, the TNT is passing SIP invite to the
2013 Aug 18
1
Asterisk SIP Trunk between two Asterisk Servers
Hi,
Am making a simple SIP trunk between two Asterisk server,
Server 1
sip.conf
[usman02]
type=peer
username=usman02
secret=usman02
host=10.30.2.58
context=man02-trunk
port=5060
qualify=yes
disallow=all
;allow=g729
allow=g729
;allow=alaw
nat=force_rport,comedia
dtmfmode=rfc2833
relaxdtmf=yes
insecure=invite,port
extensions.conf
[man02-trunk]
exten => _1X.,1,Dial(SIP/usman02/${EXTEN})
exten
2005 Oct 10
1
2 line SIP ATAs with Asterisk using RealTime
I am running CVS Head i686 running Linux on 2005-06-30 22:55:14. I have
SIP Buddies installed using MySQL.
If I try to set up a ATA that has 2 two phone lines (resulting in 2
lines on 1 IP address), my second line can never authenticate to dial out.
I ran ethereal and found that Asterisk is "looking at the IP the request
came from" and then, apparently looking up the IP address in
2007 Feb 08
0
SIP Re-Invite behind a NAT
SetUp:
- Asterisk behind a NAT,
- Red Hat 9.0
- Asterisk 1.2.14
My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have
my dial plan set up so that when outside callers dial the DiD, the
call is answered by my auto-attendant. The caller can then select who
they'd like to speak to and the call is transferred to the external
line associated with that person (usually a mobile
2014 Nov 10
0
Asterisk 12.7.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 12.7.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs
2014 Nov 10
0
Asterisk 12.7.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 12.7.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs
2011 Nov 17
0
2 same sip extension number on 2 asterisk - call not passing on certain condition
Hi list,
something crazy here. 2 asterisk on 2 different place (1.4 and 1.8) both
having an extension [115], one as type peer (caller side 1.4) and one as
friend (callee side 1.8). Phones from both location connect to Asterisk
from LAN. Router are Linux boxes.
Connection between the 2 sites is done like this:
On the callee side
[115] ;callee
type=friend
host=dynamic
secret=otherSecret