[asterisk-users] Directory Application

Hello all, Anyone know an easy way to have the 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:

[asterisk-users] Confbridge

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

Re: [asterisk-users] ICE error

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

[asterisk-users] includes with time and timezone.

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

[asterisk-users] PJSIP

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: I have tried a lot of changes to get to this but nothing works. I am getting this From: sip:109643...@xaccel.net;tag=42e4a9cb-59af-4d40-a21f-00261afbd3be To:

[asterisk-users] Modems

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

Re: [asterisk-users] Looking Asterisk SIP Guru

Message- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua C. Colp Sent: Thursday, June 27, 2019 10:41 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Looking Asterisk SIP Guru On Thu, Jun 27, 2019, at 11:28 AM, John T. Bittner wrote

[asterisk-users] Looking Asterisk SIP Guru

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

Re: [asterisk-users] Hacking

I took a look for that, Mysql running but blocked in the firewall. I do have a web gui but its hides the passwords + has a single login for admin with complex password. Even if they hacked the web site, they have no way of getting the passwords my configs are static in the asterisk folder. SSH

[asterisk-users] Hacking

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

Re: [asterisk-users] Fail2ban for asterisk 16 PJSIP

-boun...@lists.digium.com] On Behalf Of John T. Bittner Sent: Thursday, June 6, 2019 3:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Fail2ban for asterisk 16 PJSIP Hello Anyone have a working copy of Fail2ban asterisk filter asterisk.conf for Asterisk 16 running PJSIP. I have

[asterisk-users] Fail2ban for asterisk 16 PJSIP

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]

Re: [asterisk-users] Account code PJSIP

Hopefully, this may help someone in the future. If I set this before I dial out... it works. I have always in the past set this on hangup... that does not work anymore. John Xaccel From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John T. Bittner Sent

[asterisk-users] Account code PJSIP

Does anyone know how to set accountcode into the asterisk CDR when using PJSIP? I have tried Set(CHANNEL(accountcode)=XX) and a few other older ways... nothing works. If I add accountcode into the pjsip endpoint config it works... but I need to set it via dialplan. Any help is much

Re: [asterisk-users] PJSIP DNS ISSUE

>> wrote: Can’t you just reference everything in IPs? If not, then hardcode the IPs in your /etc/hosts file. I think that’s a bad idea, but that’s one way to ensure you always have the Ip of a domain name. From: asterisk-users mailto:asterisk-users-boun...@lists.digium.com>> On Beha

[asterisk-users] PJSIP DNS ISSUE

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

Re: [asterisk-users] opus

Does anyone know where do get opus for asterisk 16 that runs on GLIBC_2.12? 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 www.xaccel.net CONFIDENTIALITY NOTICE:

[asterisk-users] Cisco ATA 191

Hello, Anyone have a copy of the SEP or SIP config file for this type of device. ( Cisco ATA 191 ) I looked all over the internet and can't find anything. I did find the sip firmware below but very little docs on this model. ATA191.12-0-1-29.loads. Any help is much appreciated. Thanks John

[asterisk-users] BLF screen alerts

Anyone know how to turn off screen notification on phones with BLF buttons. I am using PJSIP. 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 www.xaccel.net

Re: [asterisk-users] Convert SIP to PJSIP

Hello all, I am having some trouble converting this setup from SIP to PJSIP. Any help is much appreciated. I used the converter script and get most of it but don't see a registration entry. How do you convert this entry into PJSIP. This working sip config. register =>

Re: [asterisk-users] Convert SIP to PJSIP

Hello all, I am having some trouble getting this to work under pjsip. Any help is much appreciated. I used the converter script and I see it register but can't receive or send to ringcentral. Anyone get this working with PJSIP? Works with chan_sip... This working sip config. register =>

[asterisk-users] Early or Pre VIdeo

Does anyone know if asterisk 15 supports Video before answering a call. I am running PJSIP and have tested a bunch of settings, progress, answer call before calling the other phone and even set the devices to send direct media. Source video device is a door phone that is supposed to support

[asterisk-users] CDR with conference asterisk 12

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

Re: [asterisk-users] One mailbox for multiple extensions with individual greetings

Why don't you use the voicemail copy feature? Create 3 mailboxes 1234, 6789 and 2000 for the shared. VoiceMail(1234@default2000@default,su) VoiceMail(6789@default2000@default,su) Set both 1234 and 6789 to email the voicemail to a fake email address and delete after email. A copy of the message

Re: [asterisk-users] Asterisk Real-time Static Voicemail

I am not running ODBC storage for Voicemail. Just running Real-time time static for configurations. John === ;Do you have compiled asterisk by yourself? In the Voicemail Build Option, what option have you selected? I think you need to select ODBC Storage and then

[asterisk-users] Asterisk Realtime Static Voicemail

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

Re: [asterisk-users] Hack

On 18 Oct 2013, at 04:06, John T. Bittner j...@xaccel.netmailto:j...@xaccel.net wrote: 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

[asterisk-users] Hack

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

[asterisk-users] Endpoint call forwarding

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

[asterisk-users] DTMF

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