similar to: Directory Application

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

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

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, 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

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

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 .

Anyone see this before? I have a main Asterisk box 11.4 connected to Windstream via SIP trunks in my colo. So as a did comes in they are routed to appropriate customers, in this case another asterisk 11.4 box. All is working well with the exception of DTMF. Losing the last digits so say someone hits 123... on the customer box I only get 12 This is the weird part, it only happens on 1 DID. If I

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

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

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

Anyone know how to get app_queue to send logs to MySQL or any other sql server. I found info for cdr's and even configs but nothing on queue_log. If sql is not supported in the current app_queue I will be willing to pay someone to add it. John Bittner Simlab.net

Hi! I am trying to install asterisk-0.5.0. For now I have installed libpri-0.4.0, now I'm getting the following error message when I do the make: chan_zap.c: In function `conf_add': chan_zap.c:836: `ZT_CONF_DIGITALMON' undeclared (first use in this function) chan_zap.c:836: (Each undeclared identifier is reported only once chan_zap.c:836: for each function it appears in.)

Anyone having any problems with CVS-HEAD-12/24/04-15:59:15 and ALERT_INFO I have a system setup with polycom phones configured to auto answer on internal calls. When we upgraded to the latest CVS the auto answer stopped working. My dialplan has not changed. I did a sip debug and I dont see the alert-info tag in any of the sip traces. Any help would be appreciated. Thanks John Bittner Simlab.net

Anyone know why zaptel would ignore a facility message from an ISDN PRI. I am trying to get Callerid name to work. The carrier says it on and I see it in the pri debug but asterisk never gets it. Any help would be appreciated. Thanks John Bittner Simlab.net Protocol Discriminator: Q.931 (8) len=9 > Call Ref: len= 2 (reference 572/0x23C) (Terminator) > Message type: ALERTING (1) >

Dear All I have a data.frame like that: structure(list(name = c("Gga_rs10722041", "Gga_rs10722249", "Gga_rs10722565", "Gga_rs10723082", "Gga_rs10723993", "Gga_rs10724555", "Gga_rs10726238", "Gga_rs10726461", "Gga_rs10726774", "Gga_rs10726967", "Gga_rs10727581", "Gga_rs10728004",

Anyone get the Nortel 1140E phones working with Asterisk ? These look like great phones and I would like to start using them on our deployments. I know these will work with Asterisk but the sample config files are hard to find. My next step, if I cant find anything on this list is to purchase a Nortel Communication Server for testing. If anyone has a used NCS that works with these phone via SIP