similar to: T-38 re-invite issue

>>>>> "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

>>>>> 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>

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 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

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 > >   

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 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 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 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

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

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

On 2017-04-19 11:57 AM, J Montoya or A J Stiles wrote: > I fished this out of an old extensions.conf from a defunct project. It might > be relevant to your use case: > > exten => 1571,1,NoOp(Call to 1571: voicemail retrieval) > exten => 1571,n,AGI(lookup_caller_id.agi,${CALLERID(num)}) > exten => 1571,n,NoOp(CLID is ${clid}) > exten =>

On 2020-04-01 15:12, Greg Troxel wrote: > D'Arcy Cain <darcy at VybeNetworks.com> writes: > But yet, new packets from that IP address reach asterisk. It seems > almost entirely clear to me that you have a firewall problem, not an > asterisk problem. This could well be but Asterisk is the only thing that continues to communicate. > I would test this out with a remote

We have a template for extensions and voicmail. They look like this: exten => %ACCOUNT%,1,Verbose(0,Entering extension %ACCOUNT%) same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN}) same => n,Dial(SIP/%ACCOUNT%,30) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,Verbose(0,${CALLERID(all)} going into voice mail for %ACCOUNT%)

It's probably not practical to have them answering the client's telephone! At a lot of sites, incoming calls would be handled by auto attendant, diverted to answering service, etc. --Don -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sebastian Nielsen Sent: Wednesday, May 10, 2017 2:46 PM To:

Hi All. Does anybody know if Google/Android has an API I can sign up for that will allow us to query the caller ID and find out if it is spam or a robocaller? I ask because we've had increase in spam calls and I'd like to simply play dead air or something really annoying. Thanks all, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: