similar to: Looking Asterisk SIP Guru

Hi All, I've been looking for information on how to use asterisk and early media to allow for a video-preview of the caller at the callee's phone for days... but I haven't been too successful :( I found that there seems to be a company "2N Helios IP" which claims (youtube-video) that "their" SIP server is able to provide early video (using a Grandstream 3157v2

Hello I assume I'm not the first one to think about this: Is it possible to connect an intercom and/or door bell to Asterisk, so that I can get an e-mail that someone rang my place while I was out? Even better: If used for a doctor's office, it'd be cool if patients could type their Social Security Number on a keypad, which would open the door and notify Asterisk which would then

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

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:

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 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 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 .

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

Hello developers, I noticed that no mime type for Microsoft Outlook is in the freedesktop.org.xml file. Adding the following to freedesktop.org.xml will correcly identify my *.pst files: <mime-type type="application/vnd.ms-outlook"> <comment>Microsoft Outlook</comment> <glob pattern="*.pst"/> <magic priority="50">

i recently setup a dhcp at my home and now i am getting these errors in my logs and was wondering if i set something up wrong? --------------------- dhcpd Begin ------------------------ Unknown Entries: Unable to add forward map from STEVEN-DELL.dusty.home to 192.168.1.100: timed out: 4 Time(s) Unable to add forward map from laptop.dusty.home to 192.168.1.104: timed out: 4

Hi all, I'm a newer user of markdown & MultiMarkdown, and I'm looking for the perfect setup of software for my Mac and my Android phone. I'd like to have my drafts editable from both. What I want is an attractive Mac app for writing (primarily blog posts and emails) that saves its files as .txt . Most of them save as .md or .markdown instead, and since there doesn't seem to

I have a number of machines that have 4 NICs, two of which are actually in use, running Centos 5. When they are rebooted, they seem to change the eth interface names, assigning them in different orders. I'm a little fuzzy on the details because they are at a remote location and I can't access them easily - especially after the network breaks. Shouldn't: alias eth0 bnx2 alias