similar to: DTMF

Anyone having issues with endpoint call forwarding on asterisk 11? Was working perfect with 10. Issues are not phone related have tried cisco, polycom and Xlite, all fail. Backtrack to 10 and it works ok again. Any help is appreciated. Thanks John Bittner CTO [cid:image003.png at 01CE76D7.8AB33690] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax:

Today I was hacked but caught it very quickly. This is the weird part, they hacked an IP Auth based account by simply knowing the account name. How is this possible? I am running Asterisk 11.5.0. Now it's my fault I used a dictionary based account name but how did they bypass the set ip I had under the account for this host. This also happened with fail2ban running and I pay for Humbug .

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

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:

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

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, I cannot find much on examples but I did find one in Russian that shows this to use + or - the time difference from GMT. I have been testing and it does not work. 1st question do includes work with timezone include => day,08:00-17:00,mon-fri,*,*,[+5] Not sure on the formatting, is it correct ? ... I tried without the brackets... that also doesn't work. If not supported in

Hello all, Running Asterisk 16.10.1 Does anyone know what this means? rtp_recvfrom: PJ ICE Rx error status code: 70004 'Invalid value or argument (PJ_EINVAL) How can I find what value it doesn't like ? I switched to a few different stun servers and I still get the same error. Calls still go through Any help is much appreciated. Thanks John Bittner CTO [xaccellogoemail] 380 US

Hello all, Anyone know an easy way to have the Directory Application<https://wiki.asterisk.org/wiki/display/AST/Directory+Application> lookup all the voicemail contexts in the system. Like a global option John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell: 973.390.1090

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

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

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>

Hello, Anyone see this issue, I have a conference bridge setup for a church with a Barix unit that streams audio into the bridge. The bridge is started by calling in to a number that executes a call file and the system calls the Barix unit starting the broadcast. Users then call in and can listen to the sermons live. The system works flawless with 1 issue I can't get accurate cdr's. Every

I've checked to web for the format of the .spx file, and have not found anything complete. I've attached what I have deciphered so far. There are a few question marks in my tables that I want to understand better. I've also attached the .spx file that I used in the example values. I'm trying to take Speex-encoded data from an embedded microprocessor and put it into the .spx

Hello, I've been working on this for over a week. I don't think its working the way it should. Here's what I'm trying to do: I have a Windstream dsl router with wireless > 192.168.1.100 | | Centos 6.8 server eth0 > 192.168.1.110 > gateway 192.168.1.111 eth1 > 192.168.1.111 > gateway 192.168.1.100 | | Switch > other

Now I have some access problems. This is what ends up un messages Hosts.allow is set to accept anyone on the local net. Username and passwords are the same Upsc gives proper output. System is Centos 6.8 Jul 22 17:20:06 BIFROST upscode2[10113]: Startup successful Jul 22 17:20:06 BIFROST upsd[10116]: listening on ::1 port 3493 Jul 22 17:20:06 BIFROST upsd[10116]: listening on 192.168.0.1 port 3493

Anyone have luck with pxechn32 and bootmgfw.efi? I'm getting the "Unable to retrieve first package" issue as reported by others. Really, any advice for UEFI booting into a winpe environment off of pxelinux 6.03 would be beneficial. Jason Jones Sr. Associate, Network Services | End User Computing 3947 N Oak St Ext | Valdosta, GA 31605 O +15137844955 E jason.s.jones at

Buenas tardes. Tengo un ensayo de dos factores (Variedad y dosis) Quisiera hacer anovas para cada una de las dos variedades que estoy evaluando. Lo puedo hacer cogiendo los datos primero de una variedad y luego de la otra. Pero creo que hay un comando "by variedad" que lo hace si necesidad de andar modificando el archivo de datos. Pero no sé dónde hay que colocarlo. Si alguno me puede