similar to: SRTP unprotect: authentication failure

Jonas Kellens wrote: > Question : I noticed I received an error when installing pjproject > --with-external-srtp > > I do not seems to have the srtp capability. > (However I can easily install with "yum install libsrtp-devel") > > Can this have anything to do with the no-audio-problems that I'm having ?? WebRTC requires SRTP and Asterisk has to be built with it

Hello The "res_srtp" module does not appear. How do I install it? Thanks.

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

Hi, I'm getting messages like res_srtp.c:395 ast_srtp_unprotect: SRTP unprotect failed with replay check failed (index too old), retrying == SRTP unprotect failed on SSRC 576693764 because of authentication failure 10 == SRTP unprotect failed on SSRC 576693764 because of authentication failure 160 [...] ... after a couple minutes during voice calls after which the connection is being

Hi; It seems the SNOM Phones are requesting to have SRTP but I do not have the module res_srtp. I tried to compile it with asterisk 1.8, make menuselect, but I found that it can not be used (I am not able to select it) with the following details: Secure RTP SRTP Depends on: srtp E Can use: N/A Conflicts with: N/A So, how I can use it? What I have to do to know the reason for not being able to

Hi all, I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I compiled it with SRTP support. Everything seems to work OK but I am having a weird issue. I cannot disable SRTP. I tried the /encryption=no/ in /sip.conf /and the /_SIPSRTP_CRYPTO=disable/ on my dailplan and it keeps trying to use the SRTP. Well, right now I have to have/ noload=res_srtp.so/ on my /modules.conf /otherwise

Hello! I have just upgraded to asterisk 1.8.2.1 and see some weird messages in log when client tries to register: [2011-01-19 00:52:47] WARNING[25624] chan_sip.c: Failed to parse contact info [2011-01-19 00:52:50] NOTICE[25624] chan_sip.c: Peer '0010101' is now UNREACHABLE! Last qualify: 105 [2011-01-19 00:53:03] VERBOSE[25624] chan_sip.c: -- Registered SIP '0010101' at

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

I am using Asterisk 11.3.0 and just updated Nightly to 24.0a1 (2013-06017) and get a SIP 488 Not Acceptable Here response. I have no problems using the same Asterisk configuration and the same page to make a call from Chrome. I have seen other people post a similar issue, but I have not seen a solution. If someone with good knowledge of this issue were to respond with "this is a known

On Thu, Jan 16, 2020 at 11:35 AM hw <hw at gc-24.de> wrote: > On Tuesday, January 14, 2020 5:29:04 PM CET hw wrote: > > Hi, > > > > I'm getting messages like > > > > > > res_srtp.c:395 ast_srtp_unprotect: SRTP unprotect failed with replay > check > > failed (index too old), retrying == SRTP unprotect failed on SSRC > 576693764 > >

Dear all, I have a very strange problem : * external calls work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module

Full_Name: Charles J. Geyer Version: 1.8.0 OS: i686-pc-linux-gnu (Suse 8.2) Submission from: (NULL) (134.84.86.22) Two bugs (perhaps related, perhaps independent) revealed by the same Poisson regression with offset mydata <- read.table(url("http://www.stat.umn.edu/geyer/5931/mle/seeds.txt")) out.fubar <- glm(seedlings ~ burn01 + vegtype * burn02 + offset(log(totalseeds)),

Is TLS required for the usage of Spice with KVM/libvirtd? I've been through the virt-install manpage a few times now to no avail. What is wrong with my syntax here (seen below)? Thanks. ~]# virt-install --connect qemu:///system -n blahhost -r2048 --vcpus=4 --arch=x86_64 --video qxl --graphics spice,port=5931 --noautoconsole --os-type linux --accelerate --network=bridge:kickstart_br0 --hvm

Hi, (I'm not subscribed to ext3-users, please CC: me) Kernel: 2.4.18. I've just converted a 100(ish) MiB ext2 filesystem to ext3 (umount, tune2fs -j, e2fsck, mount) and it seems to be happy, except... As a test, I then filled the filesystem up, lastly copying /usr/bin (as root, so the filesystem became brim full). I then umounted it, and ran e2fsck -n -f /dev/hda6, and got the

Apologies in advance if this has come up a thousand times before but is there any way to stop this error in 1.8 ? [ Feb 26 15:09:09] ERROR[6678] chan_sip.c: No SRTP module loaded, can't setup SRTP session. -- Thanks, Phil

On Wed, Sep 2, 2015 at 1:59 PM, Leonard den Ottolander < leonard at den.ottolander.nl> wrote: > Hello Mike, > > On Wed, 2015-09-02 at 13:05 -0400, Mike - st257 wrote: > > I've been through the virt-install manpage a few times now to no avail. > > What is wrong with my syntax here (seen below)? > > > ~]# virt-install --connect qemu:///system -n blahhost

Snom default configuration is SRTP enabled. You should disable the SRTP from the phone web GUI configuration Sincerely, Sam Basan From: Mitul Limbani [mailto:mitul at enterux.in] Sent: Thursday, November 12, 2015 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] No sound with internal

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]

Hi there, Sorry to come back to that, I tried to follow the code at https://github.com/samba-team/samba/blob/master/source3/auth/auth.c#L214 (and below) but I still can't understand why one Samba client can connect, but the other can't. I can't understand why, with one client, the code would go into "check_samsec.c:183" (and return "sam_account_ok") while, with

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as