similar to: PJSIP

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

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

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

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

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 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 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

Dear all, I have samba3.0.23d running on FreeBSD 6.1. It is running with "security = ADS" and has been functioning correctly from about 3.0.14 to 3.0.22. Since I upgraded to 3.0.23(a|b|c|d) group permissions are not honoured by Samba. For example: drwxrwx--- 107 setup domain admins 3072 Nov 15 19:25 install The user setup is a windows admin user; this user can access this

I got the response below from a provider. How do I extract the Identity header and apply it to the next INVITE? Is it possible at all with PJSIP? SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 172.16.7.254:52169 ;rport=52169;received=XX.205.172.89;branch=z9hG4bK-524287-1---129f4244aaba9f04 Call-ID: 102650Mzg4NmFiNTQzOGY5NDJmNjM3OTYzNmE5MzNlZDIwZmI From: "Peter Perez" <sip:727XXX3019

I'm trying to use Wine on an Intel Mac (OS 10.4.11) to run the Amazon Kindle for PC app. I've installed Wine following the instructions at http://davidbaumgold.com/tutorials/wine-mac/ and when I do "wine KindleForPCinstaller.exe" from SH's $ prompt, it successfully creates the file "Kindle For PC.desktop" (which looks like a Windows shortcut or batch/shell

I dont know the history behind CostPerUse word so I may be missing the background associated with it. It seems that it's misnomer for what it is intended. At first sight, the word indicates that the cost is a function of uses of the register - more the uses more the cost. How do we want to define the value of CostPerUse. Should it be a function of uses? or just the target? On Sat, May 30,

Hi, I'm developping some AGI in perl (5.8.6) on i386 using Asterisk 1.0.5. I want to get some variables such as DIALSTATUS and ANSWEREDTIME after a $AGI->exec("Dial", "dial_string"); but here is what i get actually: DIALSTATUS= DIALEDTIME=ANSWER ANSWEREDTIME=18 I searched the archives and saw that $AGI->verbose could mess the access to variables, but I don't use

I have a rails 4 app that has a link_to with a post method but I keep getting: No route matches [GET] "/payments/136330" <%= link_to("Buy Now", buy_path(list_item), :method => :post) %> Ideas?? -- You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To unsubscribe from this group and stop receiving