Displaying 20 results from an estimated 9000 matches similar to: ""Proxy Authentication Required" issue"
2015 May 05
0
Authenticated SUBSCRIBE and NOTIFY's R-URI
Hello,
I've got a deployment with the SBC in between the clients and Asterisk
(11.17.1 version). When the UAC tries to subscribe for "dialog" event
package, the NOTIFY request sent by Asterisk fails.
The SBC uses a different Contact (user part) for the 1st and the 2nd
SUBSCRIBE (with Auth.).
The issue is that Asterisk sends the NOTIFY with R-URI of the first
SUBSCRIBE's Contact,
2006 Feb 01
1
Unable to Register to Asterisk through Proxy
Hi,
Has anybody come across a situation where they were unable to register with Asterisk through a SIP stateless proxy server?
I'm getting an error:
"403 Authentication user name does not match account name"
As far as I can tell the requests reaching Asterisk with and without the proxy are identical excepting the IP address the REGISTER request is coming from and the Via header
2005 Aug 02
1
stale nonce
I just updated one of my stable asterisk systems to head to test it
out.. and I'm receiving a interesting log message now in asterisk..
Aug 2 13:20:56 NOTICE[15382]: chan_sip.c:5617 check_auth: stale nonce
received from '<sip:3034585725@voip.livewirenet.com;user=phone>'
(one line per registration)
I'm using an AudioCodes mp108.. it worked fine with the latest stable..
2003 May 15
8
SIP behind NAT (*sigh*)
Hi guys,
sorry to be iterating this on the list once more, but I'm not able to get
this stuff to work as I'd expect. So far, I've always managed to keep it
out of NAT environments :->
My home LAN is NATed by a simple Draytek router.
In the home LAN is an ATA186 with SIP. On the internet (public) is an
Asterisk server.
I have nat=yes in the sip.conf and the connectmode is set
2007 Aug 15
1
why is nonce="584760da" used in sip packets?
Hi all,
There is a parameter called "nonce" included in every register request that
a UA sends to asterisk. I have read sip debug a lot and only found out that
the "nonce" parameter value which is used in register request was generated
by asterisk server in a previous sip response.
As you can see in the sip debug (labled in red).
<--- Transmitting (NAT) to
2009 Nov 24
2
can't get pap2 to register from outside the LAN.
I am having a hell of a problem trying to get a linksys pap2t to
register with my asterisk from outside the LAN.
I have tried every combination of NAT, outbound proxy, stun, specify
external IP address etc and it just won't work. Here are the relevant
details.
In asterisk I have set the following.
externip=my.ip.address
localnet=192.168.0.0/255.255.0.0
nat=yes
bindport=5060
here is the
2004 Sep 08
4
WellGate 3504A with Asterisk SIP authentication and config
hey * folk,
am trying to configure a WellGate 3504A FXS SIP ATA
(http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set
up two SIP clients in sip.conf as follows:
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP channel to
context = default ; Default context for incoming calls
[1235]
host = dynamic
secret = somepass
context = default
type
2013 Apr 09
1
[OpenSIPS-Users] 404 When BYE initiated by external callee
On Tue, Apr 9, 2013 at 1:22 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>wrote:
> **
> Hi Nick,
>
> The BYE is not properly formed and rejected by script - in the 200 OK of
> the INVITE, you can see that your opensips is doing Record-Routing, but the
> BYE does not contain the corresponding Route hdr, so SIP routing is
> impossible.
>
> Regards,
>
>
2009 May 22
3
No response to our critical packet problem
Hi,
I have a strange problem. At a site where there are 20+ phones, there
is one phone that cannot make outbound (to PSTN) calls.
Each call is dropped after 20s with "no response to our critical packet".
Calls to voicemail and internal extensions work fine.
I understand that everything points to a NAT problem, but I don't
understand how it could be because:
1) It does not affect
2020 Mar 23
2
Attempting to get BLF working with linphone
So I've got a bit further with my project to get BLF working between
asterisk and linphone.
Initially asterisk was rejecting linphone's SUBSCRIBE messages because
they didn't have an Accept: header. I've fixed that and now the initial
SUBSCRIBE messages work and I see all my online contacts in green.
But after a few minutes linphone attempts to renew the subscriptions and
2011 Nov 22
1
Asterisk refuses INVITE (401) and I don't know why
Hello list,
this is the communication between an Aastra 5000 PBX and Asterisk, where
the Aastra makes a call to Asterisk. For some reason, Asterisk responds
with 401-Unauthorized and I don't know why.
Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT
with this Aastra.
A1.A1.A1.A1 = IP-address Asterisk PBX
AS.AS.AS.AS = IP-address Aastra PBX
Aastra PBX makes a call
2005 May 09
3
Zyxel 2000W (WI-FI) Problems
Hi!
Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing.
It works fine if I call the 2000W from other phones.
I have tried many sip settings. I use this now:
[205]
type=friend
username=205
secret=passwd205
callerid="Zyxel" <205>
host=dynamic
context=local
nat=yes
canreinvite=no
disallow=all
allow=g729
2004 Nov 29
2
SPA-2000 Dropped calls
Been having a problem with my two Sipura 2000's dropping calls from the
SPA-2000 side. Seems the calls are dropped right before the "Next
Registration" time. Calls drop about ever 60 minutes or so. I have
dialed from one port to the other and let it sit. After about 60 minutes
or so the calls get dropped.
System details are below
Asterisk ver. CVS-HEAD-11/27/04-23:42:45
RHEL 3
2011 Feb 10
2
Unable to make outgoing calls with Internode
Surely there must be someone here who can help me with this problem.
I have spent weeks trying to get this damned service to work with no
luck. I have incoming calls working, but no outgoing. If get outgoing
working then incoming don't work.
I have sent this problem to this list a couple of times with little or
no response, and I _really_ need some help to sort it out.
I have an asterisk
2005 Jul 24
2
Why can't sip/200 call sip/202
I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get:
sip show peers
Name/username Host Dyn Nat ACL Mask Port Status
202/202 192.168.0.6 D 255.255.255.255 5060 Unmonitored
201/201 (Unspecified) D 255.255.255.255 5060 Unmonitored
200/200 192.168.0.3 D 255.255.255.255 5060
2004 May 18
0
No luck using asterisk as proxy...
Still no luck using asterisk as a proxy.
48 hours solid working on this. I'm beginning to think asterisk isn't
going
to be compatible with the provider I'm using :(
Has anyone got *any* clues as to what can cause this message? It's
definately
provider specific (voiptalk works, pipecall doesn't) but confusingly
seems to
be caused by something in the client phone app.
I
2015 Mar 16
1
Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
On Sun, Mar 15, 2015 at 5:56 PM, Sonny Rajagopalan
<sonny.rajagopalan at gmail.com> wrote:
> George,
>
> I have the detailed log below. (Resending after trimming the email to 40KB.)
>
> The sequence below just repeats ad-nauseam. Is this a SIP trunk issue?
>
> Thanks!
>
I don't see anything obvious. The registration works though, right?
You might want to compare
2004 May 04
4
mediatrix 1104
Hi all,
I just got a Mediatrix 1104 evaluation unit -- a 4 port fxs sip gateway.
There's no printed documentation shipped with the unit, but I have a piece
of software for windows that shipped with a different model (which I haven't
tried configuring yet), that uses snmp to set misc variables (ip settings,
sip stuff, etc.). Fairly baroque interface & pretty slim on help...
2020 Mar 23
3
Attempting to get BLF working with linphone
On 23/03/2020 18:51, Joshua C. Colp wrote:
> On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com
> <mailto:john at calva.com>> wrote:
>
>
>
> Why is asterisk giving an error 500? I can find no reason, there
> is nothing in any log.
>
>
> The sequence number is from the past. The first SUBSCRIBE is sequence
> number 22 (check the
2005 Jul 05
1
Stale nonce received?
The use of the nonce looks right to me. Can somebody point out what
is going wrong here?
Jul 4 14:37:04 VERBOSE[12919]:
Sip read:
REGISTER sip:voip.victoria.tc.ca SIP/2.0
Via: SIP/2.0/UDP 199.60.222.229:5060;branch=z9hG4bKUF2CCx
Max-Forwards: 70
To: pcnavideo <sip:pcnavideo@voip.victoria.tc.ca>
From: pcnavideo <sip:pcnavideo@voip.victoria.tc.ca>;tag=w23oUOnv7kR
Call-ID: