Displaying 20 results from an estimated 100 matches similar to: "Asterisk 11 SRTP: unsupported crypto parameters: UNENCRYPTED_SRTCP"
2009 Oct 02
0
srtp issue
Hi,
I have set up an asterisk with TLS and SRTP support. The SRTP is working
with Phonerlite softphone. I have problem with the SRTP, when I make calls
on Audiocodes gateway . I got the folloowing messages on asterisk:
[Oct 2 10:59:48] NOTICE[24868]: sdp_crypto.c:232 sdp_crypto_process: Crypto
life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80
2010 Dec 24
5
SRTP unprotect: authentication failure
Hello!
Ater several successful SRTP-enabled calls with SRTP set to Mandatory, asterisk starts to give the following warnings in Log:
WARNING[13714] res_srtp.c: SRTP unprotect: authentication failure (continiously)
and client hears no sound. After i restart the client program it works fine again for a while. Then the same warning again.
Asterisk 1.8.1.1, RealTime engine, sip peer has
2014 Oct 09
1
sdp_crypto_process: Crypto life time unsupported: crypto
Hello,
I have added the following to the peer definition :
ignorecryptolifetime=yes
But still Asterisk tells me :
[Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:244
sdp_crypto_process: Crypto life time unsupported: crypto:1
AES_CM_128_HMAC_SHA1_80 inline:ikW6yFvdVkSaeTuVO1isTQkdaxOjgQjMEMSGUf+K|2^32
[Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:254
sdp_crypto_process: SRTP crypto
2014 Oct 07
1
Grandstream GXP2160 + SRTP
Hello,
I am trying to setup a Grandstream GXP2160 IP-phone with secure calling
(SRTP).
Secure signaling SSIP for registration is working great !
I follow this guide :
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
But when I try to make a call with SRTP, I get stuck. There is an
initial INVITE which is anwered with a 401. There should follow a new
INVITE with a nonce,
2007 Jul 31
3
Nonlinear optimization with constraints
Hello R community,
I am using R for creating a model using optimization. I would like to ask if there is R-function/package for solving the problem below:
Minimize sum(abs(exp^(Ai1 x1 + Ai2 x2 + ... + Aim xm - bi) - 1)), for each i = 1, ..., n.
subject to Ai1 x1 + Ai2 x2 + ... + Ajm xm - bi <= c, where c is a scalar.
(x is a vector of variables, A is nxm matrix, b is a vector)
2015 Mar 04
0
TLS connect() error when calling udp to tls
Stuck with TLS transport,
I have 2 phones (both in local network for tests)
one connected with up second with tls
when I calling TLS to UPD everything is fine, but when UDP calls TLS I getting an error
ERROR[44230]: pjsip:0 <?>: tlsc0x7f143012 TLS connect() error: Connection refused [code=120111]
pjsip log:
-- Called PJSIP/601/sip:601 at 192.168.1.55:5075;transport=tls
<---
2013 Jul 03
1
Custom dial plan for internal transfers of external calls
Arch = x86_64
OS = CentOS-6.4 (freepbx)
Asterisk = 11.4.0
FreePBX = 2.11.0.2
We use Snom870 handsets with firmware v.8.7.3.19.
I am trying to develop a custom dial plan to invoke a distinctive
ring-tone when an external call is transferred internally. Based on
an earlier solution I discovered I am attempting this:
[from-internal]
include => set-alert-if-local
[from-internal-original]
2015 Apr 24
0
Real sh? Or other efficient shell for non-interactive scripts
On Fri, Apr 24, 2015 at 09:47:24AM -0700, Gordon Messmer wrote:
> On 04/24/2015 03:57 AM, Pete Geenhuizen wrote:
> >if you leave it out the script will run in whatever environment it
> >currently is in.
>
> I'm reasonably certain that a script with no shebang will run with
> /bin/sh. I interpret your statement to mean that if a user is using ksh
"It
2006 Dec 06
0
Error in codec string '=audio 5004 RTP/SAVP 3'
Hello,
I have a problem with a grandstream IP Phone.
The SIP autentication is OK, but when try to call someone I get the message
--> WARNING[14281] chan_sip.c: Error in codec string '=audio 5004 RTP/SAVP
3'
I tried to change the CODECs (ulaw, alaw, GSM, etc), the result is always
the same.
Tried to change the RTP port but the result is the same.
The grandstream IPhone is behind a
2012 Jun 20
1
Overview of SIP error codes and possible causes?
Hello,
is there anywhere an overview of SIP error codes and under which condition
they are reported by Asterisk?
There are general definitions for SIP error codes, but they are quite
general and it's Asterisk that actually checks what's wrong and then
reports an error. Now, currently I could check the source code to get more
informations what could have caused the error, but that's
2015 Mar 12
2
WebRTC demo phones
Hello,
Can anyone recommend a particular online WebRTC phone for testing with
Asterisk?
We tried:
- JsSIP, but even with the "enable video" checkbox disabled it sends video
options in the INVITE SDP and Asterisk rejects it with "Rejecting secure
video stream without encryption details".
- sipML5, but it won't register, perhaps something to do with not using the
Asterisk
2014 May 09
1
deactivate SRTP in asterisk 11
Hi all,
i try to deactivate SRTP in asterisk 11.
In sip.conf:
tlsenable=no
encryption=no
transport=udp
srtpcapable=no
but when I try to make a call comes following message:
[May 9 15:19:03] DEBUG[24745][C-00000086]: sip/sdp_crypto.c:285 sdp_crypto_process: Accepting crypto tag 1
[May 9 15:19:03] DEBUG[24745][C-00000086]: sip/sdp_crypto.c:310 sdp_crypto_offer: Crypto line: a=crypto:1
2012 Jun 18
1
Error SIP/2.0 488 Not acceptable here
Hello,
a person trying to call me by my phone number is getting the error 488 Not
acceptable here. I googled that error, seems like this error is normally
caused by a failed codec negotation, though I have no clue how I could have
read this out of the logs. Anyway, my setup is as follows:
Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider
The user calling me is also using Sipgate and is calling my
2012 Dec 17
1
[webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
Dear All,
I use sipml5 to register two users from browser and the two clients are
successfully connected. But when I made a call from one of the users, the
other user doen'st have call notification and for a while the calling
process ended. I check the /var/log/asterisk/messages got the following log:
[Dec 17 14:54:11] WARNING[11471][C-00000000] chan_sip.c: Received SAVPF
profle in audio
2016 May 30
2
Need stronger SRTP ciphers (256 bit)
Hi folks,
At least several endpoints (soft phone and desk phones) are supporting various 256 bit ciphers for SRTP these days. I *believe* libsrtp has been updated to allow this, and that only the code in Asterisk has not been been updated to allow these stronger ciphers.
Would anyone with the know-how be willing/able to submit a patch ?
Thank you,
Kevin Long
2011 Aug 03
2
snom and srtp
Hi,
I am running asterisk 1.8.5.0 and have compiled in the srtp module
All but Snom phones are working.
I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when trying to make a call (this is snom calling snom).
---------snip------------------
== Using SIP RTP CoS mark 5
-- Executing [10000 at
2014 Mar 14
0
sipML5, Ast12 and WebRTC: not acceptable here
Hi All.
I'm running some tests with the latest Asterisk SVN-branch-12-r410493M
compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS
machine (2.6.32-358.18.1.el6.i686).
As a client I'm using the sipMLP WebRTC javascript softphone running on
Chrome 33.0.1750.146 m.
I have the softphone correctly registered on the Asterisk machine but as
soon as I try to start a new call
2012 Oct 04
49
[RFC 00/14] arm: implement ballooning and privcmd foreign mappings based on x86 PVH
This series implements ballooning for Xen on ARM and builds and Mukesh''s
PVH privcmd stuff to implement foreign page mapping on ARM, replacing
the old "HACK: initial (very hacky) XENMAPSPACE_gmfn_foreign" patch.
The baseline is a bit complex, it is basically Stefano''s xenarm-forlinus
branch (commit bbd6eb29214e) merged with Konrad''s linux-next-pvh branch
2014 Jul 10
0
Asterisk 1.8.29.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.29.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.29.0 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 Jul 10
0
Asterisk 1.8.29.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.29.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.29.0 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