similar to: Asterisk Real-time Static Voicemail

Anyone know how to disable DNS in asterisk so PJSIP still works when the internet goes down. I tried a few things but nothing is working. I even installed BIND on the asterisk box ...that didn't even work. Once I pull the plug on the internet, I cant dial anything. John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax:

Hello Anyone have a working copy of Fail2ban asterisk filter asterisk.conf for Asterisk 16 running PJSIP. I have tried 10 different filters but none of them show any matches when testing with fail2ban-regex I see date template hits but no matches.... My log [2019-06-06 15:37:20] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" <sip:2405 at

Hello, I am looking for a consultant that know asterisk in and out including how to troubleshoot sip and rtp. I have a device that this acting very strange and I need to prove it's the device code and not an issue with my setup. Very simple setup, all local no nat... Grandstream video phone and a AIphone IX-MX7 door station. PJSIP ... doorstation to grandstream 3370 works perfectly. Early

Anyone know how someone can hack an asterisk box and register with every single account on the box. This box only has 3 accounts, with very complex passwords. Have VoIP blacklist setup and fail2ban... The hackers were able to make 2 calls to Cuba before my alerting system texted me. I am running asterisk 16.3 with PJSIP. This is my only box open to the outside world, a requirement for this one

Guys, I need you help on this one. Don't know when this broke but we have a custom gui that runs on top of Asterisk running a real-time static for configurations. Nothing has changed with the database other than upgrades of Asterisk 10. Customer complained that there password was not changing when they called into voicemail and changed it. Database is running standard ast_config with the

Hi, Is there a way in Asterisk 11 to use a single voicemailbox for multiple extensions while still hearing each extension's individual greeting? Use case: someone has 2 numbers and wants all voicemail messages for both numbers to end up in one mailbox. So when dialing 1234 and NOANSWER you would hear "the person at extension 1234 is unavailable" and the message would be stored

Joshua, Regarding Outbound Registrations in realtime and a reload. Does it require a "pjsip reload" or full asterisk restart? Antonio G?mez Soto wrote: > Hi, > > The asterisk wiki page says: > > "Sorcery.conf allows you to try to configure other PJSIP objects such > as transport using realtime and it currently won't stop you from doing so. > However,

Excellent. I was using ast-13.1.0 with no luck. I upgraded to 13.2.0 and have made it further, but am having a little difficulty. The outbound-publish object types seems to be working in realtime now. But the asterisk-publication object is only reading from sorcery.conf. I know you said that it *should* work, with no guarantee, which I'm fine with. I just want to make sure I don't

Any ideas? This one would make the 10th email received. Return-Path: <centos-bounces at centos.org> Received: from psmtp.com (exprod8mx192.postini.com [64.18.3.171]) by mail.pdinc.us (8.12.11.20060308/8.12.11) with SMTP id p2LKkur4025229 for <jpyeron at pdinc.us>; Mon, 21 Mar 2011 16:46:57 -0400 Received: from source ([72.26.200.202]) by exprod8mx192.postini.com ([64.18.7.10]) with

We have a mail filter ( CanIt, by Roaring Penguin, see http://www.roaringpenguin.com ) that is doing a very good job indeed of blocking spam and viruses. And I certainly get a lot from this list ... so much that without the use of CanIt, I would have already had to retire my email address, and possibly even our entire domain, to avoid the constant march of crap. But ... the list server keeps

Hi I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to the caller on pick up the is an option A(x) where x is the file to play to the called party. Also

Hi all, We have recently had some power failures have led to corrupt mdbox instances. Dovecot?s fsck appears to get caught in a loop when the recipient has a large amount of mail. In the example log output below, each ?rebuilding indexes? seems to be interrupted by a new IMAP connection. Later, when our folks realized the issue and blocked access to the mailbox, the index rebuild finished in

Hello, I'm just trying to setup two mailservers with dovecot 2.2, postfix, mysql, postfixadmin where i would like to sync the maildirs between them; I'm at the point where sending/receiving and qouta is working and try the maildir replication but it only creates the maildir directories (as example: /srv/vmail/test.domain/screenie/) on the second server but nothing inside the maildir is

Hi everybody, I'm trying my Asterisk to send emails when a new message arribes to a voicemail user but no email arribes. my voicemail configuration is the following: VOICEMAIL.CONF: [general] format=wav serveremail=asterisk at mydomain.com attach=yes maxmsg=20 maxsecs=180 minsecs=3 maxsilence=10 silencethreshold=128 maxlogins=3 fromstring="My Asterisk" When I look at maillog file,

Guys, I have a customer that heavily uses modems, the problem they don't work reliably with some of the carriers I have used like Level3. This is somewhat expected due to the limits in VoIP so I need a better solution. If I set up an asterisk system on customer premise with an FXS card in it and have calls sent to another asterisk box with a PRI can I get this to be more reliable and better

Hi, all -- I''m in the process of configuring a new machine for use as a Xen server, and am having some rather significant stability issues. The hardware/distro/kernel info is: opensuse 10.3 Linux offxen2 2.6.22.5-31-xen #1 SMP 2007/09/21 22:29:00 UTC x86_64 x86_64 x86_64 GNU/Linux 2 x Dual-Core AMD Opteron(tm) Processor 2212 HE 8 x 2GB PC2-5300 RAM Xen 3.1.0_15042-51 I currently have

To all: No matter what I try, I cannot get the system to wait for the admin to join. It just dumps users into the bridge directly. I do not have a pin for users, does that matter? What am I missing? Another issue the absolute timeout is not working ? ... have recordings that last for over 24 hours... and this should not happen... All calls should hangup after 4 ? Any ideas ? Any help is much

Hello, Anyone know how to set the "To:" in an invite for PJSIP to custom settings. I got the "from" to be the way I need it. From: <sip:e04f43a2ed59 at xaccel.net;tag=44l1nRmW2 To: "TEST" <sip:5tf2f2s0rbtdj-20d14fl6n65t0o-0u03 at 34.221.174.202> I have tried a lot of changes to get to this but nothing works. I am getting this From: sip:109643183 at

Hello, have you noticed the message num (VM_MSGNUM) is off by one? For example, I receive the following message: "Just wanted to let you know you were just left a 0:03 long message (number 7)" but in attach there is the msg0006.wav Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have two mail servers and am also experiencing sporadic replication errors over tcps, similar to Reuben. Each server is running Dovecot 2.3.11.3 (502c39af9) on Debian 10.6. *Log entries from MX1* Nov 18 00:39:26 mx1 dovecot: dsync-local(user at example.com)<Ow3zAjWxtF+TDgAAPHKnuQ>: Error: dsync(mx2.example.com): I/O has stalled, no activity for 600 seconds (last sent=mailbox,