similar to: Permission denied to access the email file

Greetings all, First time poster, Sorry if this has been answered here before. We recently replaced a failed 1.4x asterisk PBX at a customer location. Voicemail access was setup when the customer dialed *8, This worked in 1.4. Now, Running 1.6 (I know it's old I had to load it quickly, And that's what I got working first. It'll get upgraded to 1.8 soon). The strange part is *8 no

Calling my home numbers has always worked. Till now. The Asterisk CLI show the following : [Dec 23 10:53:22] NOTICE[25159]: chan_sip.c:12640 handle_response_invite: Failed to authenticate on INVITE to '<sip:092xx90xx at 85.xx.xx.xx>;tag=as5b139383' And after restarting Asterisk, the CLI is flooded by : [Dec 23 11:11:06] WARNING[1468]: chan_sip.c:1817 __sip_xmit: sip_xmit of

I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001 at 192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes.

Hello, I have a problem with the import of a date file. I seems verry tricky. I have a text file (end of the mail). Every file has a different number of measurments witch start with "START OF HEIGHT DATA" and ende with "END OF HEIGHT DATA". I imported the file in a matrix but the letters before the numbers are my problem (S= ,S=,x=,y=). Because through the letters and the

Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten => 10,1,NoOp() exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric" <sip:10 at 192.168.1.150>;party=called ) exten => 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12

Hello, Seeing a strange thing with dovecot 1.0 b7 pop3. If a user gets a particular _very_ short spam message (not sure what virus makes these..), with NO Subject and NO To: headers and NO body, (example available upon request), the user is unable to download the mail. The log entry is: May 10 10:51:35 mail dovecot: POP3(user): Disconnected top=0/0, retr=1/1344, del=0/75, size=16383381

The following bugs look like they only relate to llvm-gcc. Can they be closed, as llvm-gcc is no longer supported? http://llvm.org/bugs/show_bug.cgi?id=3636 http://llvm.org/bugs/show_bug.cgi?id=5011 http://llvm.org/bugs/show_bug.cgi?id=6764 http://llvm.org/bugs/show_bug.cgi?id=8451 http://llvm.org/bugs/show_bug.cgi?id=9310 http://llvm.org/bugs/show_bug.cgi?id=9311

Hi all: I am using the libvirt version that comes with Ubuntu 18.04.3 LTS. I have written a script that backs up my virtual machines every night. I want to limit the amount of memory that this backup operation consumes, mainly to prevent page cache thrashing. I have described the Linux page cache thrashing issue in detail here:

On 1/22/20 11:11 AM, Michal Privoznik wrote: > On 1/22/20 10:03 AM, R. Diez wrote: >> Hi all: >> >> I am using the libvirt version that comes with Ubuntu 18.04.3 LTS. > > I'm sorry, I don't have Ubuntu installed anywhere to look the version > up. Can you run 'virsh version' to find it out for me please? Nevermind, I've managed to reproduce with

Hi Jay, > The following bugs look like they only relate to llvm-gcc. Can they be closed, > as llvm-gcc is no longer supported? > > http://llvm.org/bugs/show_bug.cgi?id=3636 > http://llvm.org/bugs/show_bug.cgi?id=5011 > http://llvm.org/bugs/show_bug.cgi?id=6764 > http://llvm.org/bugs/show_bug.cgi?id=8451 > http://llvm.org/bugs/show_bug.cgi?id=9310 >

Hi, I'm stucked in situation, and look for a work around if possible in Asterisk. I have a dialplan, [default] exten => 111222,n,Set(fu_callerid=141688xyxzz) exten => _X.,n,NoOp(Callerid ${fu_callerid}) exten => _X.,n,wait(2) exten => _X.,n,Answer() ? When, ?Answer Application is called AMI Event is triggered like this.. ? ? ? ? ? 'Event' => 'Newexten', ? ? ? ?

Hi All, I am using rsync to backup our office server to our Internet server (RHE). As an association for doctors we are looking at providing a backup service for their practices using rsync. As it would be patient data it would need to be encrypted. I have found a few options, namely esync wurt rsyncrypto Does anyone have experience with the above and perhaps like to recommend one? On the

https://bugs.freedesktop.org/show_bug.cgi?id=99954 Bug ID: 99954 Summary: Errors when using VirtualBox with 3D acceleration: gr: ILLEGAL_CLASS ch 6 [007f7f8000 VirtualBox[5063]] subc 0 class c000 mthd 2390 data 00000000 Product: xorg Version: unspecified Hardware: Other OS: All

Hello, I have asterisk setup with Broadvoice. It works great as PBX and Outgoing calling server for all local clients withing 192.168.1.0 network. My asterisk is running over NAT. I use linksys router. Now, I am trying to connect from outside to my asterisk server. I use Diax as iax client. For some reason I cannot connect to my server from outside. On my router I forward those ports to my

Hello list. My phone rings, I pick up, and the conversation is terminated. Every time. The setup : Grandstream GXP2010 --> SIPproxy (Endian Firewall) --> Asterisk Server --> ITSP Could it be the SIP proxy of my Endian firewall ?? I have 4 accounts on the Grandstream which listen on port 5060 --> 5063. They have a proxy defined namely my Endian firewall. On this SIPproxy I have a

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

Hi I have a big problems on my Asterisk systems : I have one Asterisk Server with static IP (no nat) and 6 Linksys SPA941. All SPA are after a router with NAT: * SPA-1 and SPA-2 are on the same network, we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet router * SPA-3, we have a pat 5062 => SPA-3 * SPA-4, we have a pat 5063 =>

(oops accidentally sent to -owners) Hi, i'm trying to start a LXC guest on a F14 computer .. followed the examples in http://libvirt.org/drvlxc.html, but got stuck when starting it .. 03:07:23.706: debug : virCgroupNew:542 : New group / 03:07:23.707: debug : lxcControllerRun:563 : Setting up private /dev/pts 03:07:23.711: debug : lxcControllerRun:589 : Mouting 'devpts' on

I just finished installing Asterisk 13 on our test server and I can now use PJSIP to register phones and make and receive calls. The only problem I am having is that when I register multiple phones to a single account only one of them rings. The AOR for the account has maxcontacts at 3. If I do a pjsip show endpoints I can see two "Contact" entries which I take to mean that

> Joshua Colp wrote: > > There should be nothing different, except for how you configure things. > What is the full PJSIP configuration? What is the environment where > Asterisk is running? Is ICE actually in use on the other side? What is > the full SIP trace? > The full configuration is here: http://pastebin.com/XqZG1m5X I am connection over TLS / SRTP on port 5063. When