similar to: Res_Srtp

> exten => _13XXXXXXX,1,dial(${OPERATOR},20) Hello "SIP/2.0 401 Unauthorized" Unfortunately the negative. An asterisk indicates a 404 error. On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle <support at drdos.info> wrote: > > On 3/5/19 2:46 AM, Gokan Atmaca wrote: > > Asterisk can send calls, but I don't get a call. What could be the problem? > > >

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

> Show us the parts of your dial plan which: > > a) work for recording incoming / outgoing calls > > b) do not work for recording internal calls > > then we might be able to give you a clue what's wrong. Hello For example: My phone number is 1000, the other's number is 1001. These numbers are in the same PBX (asterisk). I want 1000, 1001 can call to record audio

Hello Asterisk can send calls, but I don't get a call. What could be the problem? [from-siptrunk] exten => 13XXXXXXX,1,dial(${OPERATOR},20) Thanks.

Hello Mynum: 6001 , Othernum: 6002. I can record as follows. But I do not enter individual records for each internal required. I want to do it more smoothly with a Macro. Thanks. exten => _6001,1,NoOp() exten => _6001,n,MixMonitor(${UNIQUEID}.wav,ab) exten => _6001,n,Dial(SIP/6001,20) exten => _6001,n,StopMixMonitor() exten => _6001,n,Hangup() On Sat, Mar 9, 2019 at 6:50 PM

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

> If you run top what are you seeing on the %Cpu(s) line? %20 On Mon, Feb 8, 2016 at 9:30 PM, Alvin Starr <alvin at netvel.net> wrote: > Slow disks will show up as higher I/Owait times. > If your seeing 99% cpu usage then your likely looking at some other problem. > > If you run top what are you seeing on the %Cpu(s) line? > > > On 02/08/2016 02:20 PM, Gokan Atmaca

> I'm guessing you're using standard 7,200rpm platter drives? You'll need > to share more information about your environment in order for us to > provide useful feedback. Usually though, the answer is 'caching' and/or > 'faster disks'. Yes , 7.2k rpm disks. 2T mirror (soft). In fact, I had such a preference for slightly more capacity. Unfortunately very

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

> Please give us a link to a datasheet for that device. Hello https://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/unified-sip-phone-3905/data_sheet_c78-651588.html On Fri, Apr 12, 2019 at 3:58 PM Antony Stone <Antony.Stone at asterisk.open.source.it> wrote: > > On Friday 12 April 2019 at 14:42:57, Gokan Atmaca wrote: > > > Hello > > > >

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

Hello I use KVM. In a virtual machine "jbd2 dm-0" disk I / O is very increases. It consumes up to 99%. For this reason, slowing down the other virtual machine. What should I do to solve the problem. ? Thanks..

Hi, https://community.cisco.com/t5/ip-telephony-and-phones/cp-3905-asterisk/td-p/1995981 The phone does work, you do need to TFTP the configuration files to the phone though. Doesn't look like custom firmware is required. -- Chris. On Fri, Apr 12, 2019 at 3:29 PM Antony Stone < Antony.Stone at asterisk.open.source.it> wrote: > On Friday 12 April 2019 at 15:24:27, Gokan Atmaca

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 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

> Are the disk partitions properly aligned to 4k boundary on the host (and in > the guests too) ? > There are 5 in total server. 32G ram. 2T r1 (soft) disk. On Mon, Feb 8, 2016 at 9:41 PM, Zoltan Frombach <zoltan at frombach.com> wrote: > Are the disk partitions properly aligned to 4k boundary on the host (and in > the guests too) ? > > See >

I solved the problem. "action.d/iptables-custom.conf" include only udp. service fail2ban restart Thank you. On Sun, Sep 13, 2015 at 9:17 PM, Andres <andres at telesip.net> wrote: > On 9/13/15 11:16 AM, Gokan Atmaca wrote: >> >> Hello >> >> I'm using the Fail2ban. I configuration below. I want to try to >> prevent the continuous password.

> Please check if there is also error log contained OOM kill? It does not appear. On Wed, Dec 7, 2016 at 4:41 PM, -=X.L.O.R.D=- <xlord.sl at gmail.com> wrote: > Dear, > Please check if there is also error log contained OOM kill? > > Xlord > > -----Original Message----- > From: CentOS-virt [mailto:centos-virt-bounces at centos.org] On Behalf Of Gokan > Atmaca >

Hello How can I record voice between internalities? I can record voice in incoming and outgoing calls, but I can't make it between the internal. Would you support this? Thanks.

Hello Can I use Cisco 3950 on Asterisk ? Thanks.