similar to: Block Spam Calls

This is exactly what I do - “press 1 for a human” Works great ________________________________ From: asterisk-users <asterisk-users-bounces at lists.digium.com> on behalf of Greg Woods <greg at gregandeva.net> Sent: Thursday, December 12, 2019 6:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Block Spam Calls Most spam calls are

Hello Doug, Friday, December 13, 2019, 11:03:37 AM, you wrote: >> This is exactly what I do - “press 1 for a human” >> Works great > I do this as well, but I also do a database lookup to see if the number > is on our speeddial list and if so, pass the call directly on without > the IVR prompts. I do something similar for calls without caller ID, but I was still getting

>>>>> D'Arcy Cain <darcy at VybeNetworks.com> writes: >> Ie after both sides select t38, until they agree on the t38 terms. > OK, so does that mean that setting it to 25000 should leave time for the > re-invite or does the timeout start after that. As I wrote above, after that. After the sip/sdp. -JimC -- James Cloos <cloos at jhcloos.com>

Hi Rowland, thank you for your effort! However, this is entirely not what I am trying to achieve. What I am trying to achieve is to get the "prexec" method to work. The reason behind this is that I would like to have a zfs data set created per user in an automatic (scripted) way. The reason behind that is that if I do this by hand - from a domain admin account and with the ACL

I have an error sending to a specific fax number. It may be more than one but this is the one I investigated. It seems the delay for the SIP negotiation in T.38 was initiated after 6 seconds, however, our system sent the BYE after only 4 seconds, possibly cutting the call before all the communication necessary for the negotiation was completed. Here is the trace from our provider showing their

On 2/04/2020 6:35 AM, D'Arcy Cain wrote: > On 2020-04-01 16:28, Mark Boyce wrote: >> On 1 Apr 2020, at 22:14, Greg Troxel <gdt at lexort.com >> <mailto:gdt at lexort.com>> wrote: >>> I think you need to use tcpdump and turn up firewall debugging. >> sngrep is your friend …My bet is UDP vs TCP on firewall rules :-) > block drop in log quick on bge0

I am running Asterisk 16.9 on FreeBSD 12.1-RELEASE-p1. I keep seeing lines like this in my logs. [Apr 1 13:30:33] NOTICE[101155][C-00004526] chan_sip.c: Call from '' (45.143.220.235:5356) to extension '2037' rejected because extension not found in context 'unauthenticated'. I have a script that checks for things like this and adds them to my packet filter (pf).

On 1 Apr 2020, at 22:14, Greg Troxel <gdt at lexort.com> wrote: > > I think you need to use tcpdump and turn up firewall debugging. sngrep is your friend …My bet is UDP vs TCP on firewall rules :-) Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 2017-04-18 02:42 AM, Pete Mundy wrote: > Try this: > > asterisk -r > core set verbose 10 > [get user to trigger fault] > [examine console output, and post to list if still unclear] > > If you don't solve it yourself, then we'll be able to help further once > we've seen the output. I can't see much more than at my previous debug level but here it is

>>>>> "DC" == D'Arcy Cain <darcy at VybeNetworks.com> writes: DC> Perhaps someone can explain what t38timeout is supposed to do A 'git grep t38timeout' on the src leads one to res/res_fax.c, where one case see that it is the number of miliseconds to permit for t38 negotiation to complete once it starts. Ie after both sides select t38, until they

I am having troubles with sending faxes. I hope someone can help me work out a better method. Basically we have a special address that our users can send to. It winds up on our Asterisk server which runs a Python script that parses the message for attachments and the phone number from the recipient address. The attachments are converted to TIFF and stored in a folder with various information

On 2017-04-18 08:17 PM, Pete Mundy wrote: >> On 19/04/2017, at 7:58 am, D'Arcy Cain <darcy at VybeNetworks.com >> <mailto:darcy at VybeNetworks.com>> wrote: >> >> <snip> >> Everything looks the same as another one that works except for two >> things. The one that works doesn't have the "Probation passed" lines. >> I am

On 12/3/19 3:04 PM, Joshua C. Colp wrote: > >     The AMI command, after the login, looks like this: > > > >     Action: Originate > >     Channel: SIP/outgoing/%%(destination)s > >     Context: LocalSets > >     CallerID: Vybe Consulting Inc Fax Service <5555551212> > >     Exten: sendfax > >   

On 12/2/19 11:52 AM, Joshua C. Colp wrote: > So I know that AMI is listening and I can talk to it.  Here is the > main log" > > [Nov 27 06:16:00] VERBOSE[101155] asterisk.c: Remote UNIX connection > [Nov 27 06:16:00] VERBOSE[101245] asterisk.c: Remote UNIX connection > disconnected > [Nov 27 06:16:01] VERBOSE[101244] manager.c: Manager

I recently upgraded from Asterisk 13.19 to 16.6.1. Everything is working fine with a few minor tweaks except outgoinf fax. Incoming works fine. I do outgoing faxing through an AMI call. Here is the output from the security log: [Nov 27 06:16:05] SECURITY[101222] res_security_log.c:

On 2017-04-18 08:31 PM, Victor Villarreal wrote: > Maybe excecuting the following command at Asterisk console, will help you: > > asterisk> voicemail show users > > And you will get a list of all mailbox configured in your system. Search > for the user with problems. VoiceMail stocktrans2 Angelica Douglas 12 Definitely there. In fact, I generate all

On 2018-05-22 02:17 AM, Yves wrote: > you could > > - use "global variables" > - use the asterisk built in database Both of those seem difficult as the process is split between Asterisk and an external script. > - mv the file to temporary folder _before_ faxing (would be the most > easy solution as you already > know how to mv a file via asterisk...) True. This

Hi, all I am a pabx vendor from Singapore. Recently we are going to implement a failover solution for our customers using heartbeat, the asterisk server can failover perfectly, however the g729 codec canot work, because it is binded the mac address, we have bought two set of licenses, can you provide us some workaround for this scenario? Regards, Liangliang

On 2017-04-19 02:39 AM, Pete Mundy wrote: > Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail > show users' I can't see why the vm_authenticate function is > failing to read the username :( I can answer that one. It's because we can't enter 'stocktrans2' from a telephone so we just hang up. The question is, why does it ask for the

On 2017-04-20 05:14 AM, J Montoya or A J Stiles wrote: > This is just screaming "configuration mismatch" -- or, possibly, "latent bug > whereby things parsed in separate places should be treated the same, but are > actually getting treated differently". I really don't want to be the "my system isn't working so there must be a bug in Asterisk" guy