Re: [asterisk-users] Touch tone stutter

On 2016-11-22 07:49 PM, Pete Mundy wrote: One direction that may be worth exploring further is his ATA's config (or perhaps swapping it for a different model). Eg adjusting echo cancellation or line impedance settings. I have to be careful here as I auto-provison these devices and changes

[asterisk-users] Touch tone stutter

I am hoping someone else has seen this and can offer a solution or at least a direction to investigate. I am running 11.23. Most of my clients are fine but one has a strange behaviour. He has a Grandstream HT701 like most of my clients who use an ATA. He can make call and they are crystal

Re: [asterisk-users] Touch tone stutter

On 2016-11-27 06:46 AM, Max Grobecker wrote: Hi, you could try switching the DTMF mode of the ATA's SIP peer (and also in the ATA itself) to INBAND transmission. In this mode, the ATA doesn't need to recognise DTMF tones and your Asterisk can interpret it. For this to work, the ATA needs to

Re: [asterisk-users] Voicemail asking for login

On 2017-04-20 04:07 PM, James Cloos wrote: I enable full log and run 'core set debug 9' before doing a pair of tests. (The full log is easier to grep than the console output.) Then compare a working vs stocktrans2 side by side. I did debug 10 and saved the console output into files which I

Re: [asterisk-users] Voicemail asking for login

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

Re: [asterisk-users] Voicemail asking for login

On 2017-04-20 12:23 PM, D'Arcy Cain wrote: Here is the full dialplan for stocktrans2. I reduced this to the following and I still have the error. exten => stocktrans2,1,Verbose(0,Entering extension stocktrans2) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,

Re: [asterisk-users] Voicemail asking for login

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

Re: [asterisk-users] Voicemail asking for login

On 2017-04-20 12:52 PM, J Montoya wrote: On Thursday 20 Apr 2017, D'Arcy Cain wrote: On 2017-04-20 12:23 PM, D'Arcy Cain wrote: Here is the full dialplan for stocktrans2. I reduced this to the following and I still have the error. exten => stocktrans2,1,Verbose(0,Entering extens

[asterisk-users] Voicemail asking for login

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)

Re: [asterisk-users] PBX selection

On 2017-04-17 12:41 PM, Victor Villarreal wrote: * Asterisk is build to work on Linux. So your team needs some skills like setting up a basic Linux server (Debian, Centos, etc), donwload software from Internet, compile and install software manually. It may be that the developers mostly use

Re: [asterisk-users] Voicemail asking for login

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 =>

Re: [asterisk-users] Voicemail asking for login

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

Re: [asterisk-users] Voicemail asking for login

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

Re: [asterisk-users] Voicemail asking for login

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

Re: [asterisk-users] How to detect fake CallerID? (8xx?)

On 2017-05-10 04:15 PM, Sebastian Nielsen wrote: The thing is then to be able to record which IP is the client, but if your services are ordered by the client via some web form, you could have that IP be recorded as "client IP" and the employee must check in/check out from that IP. IPs change.

[asterisk-users] Looking for better fax handling

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

Re: [asterisk-users] Looking for better fax handling

On 2018-05-21 08:04 PM, John Kiniston wrote: > Lock files. > > Create one when you start sending the fax, on your retry process check > for a lock file and if one exists don't retry. Your suggestion is to create a lock file for each fax (there could be many concurrent ones) and have the dialplan

Re: [asterisk-users] Looking for better fax handling

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

Re: [asterisk-users] T-38 re-invite issue

On 2018-06-08 06:45 AM, D'Arcy Cain wrote: > I tried increasing the t38timeout but it did not seem to have any > effect. Is there another variable to adjust somewhere? Pinging the list to see if anyone has any thoughts. Perhaps someone can explain what t38timeout is supposed to do as it

Re: [asterisk-users] Looking for better fax handling

On 2018-05-24 02:32 AM, Yves wrote: > of course you can query asterisk asterisk and look, if your fax is still > running...: > > asterisk -rx "fax show sessions" lists you all acive fax sessions... Well, yes but how do I know which channel I am looking for? When I say "I" I mean my program. I

[asterisk-users] T-38 re-invite issue

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

Re: [asterisk-users] T-38 re-invite issue

On 2018-06-12 07:10 PM, James Cloos wrote: >>>>>> "DC" == D'Arcy Cain 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

Re: [asterisk-users] T-38 re-invite issue

On 2018-06-13 07:20 AM, James Cloos wrote: >>>>>> D'Arcy Cain 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 ti

[asterisk-users] T-38 re-invite issue

On 2018-06-08 06:45 AM, D'Arcy Cain wrote: > I tried increasing the t38timeout but it did not seem to have any > effect. Is there another variable to adjust somewhere? Pinging the list to see if anyone has any thoughts. Perhaps someone can explain what t38timeout is supposed to do as it

Re: [asterisk-users] T-38 re-invite issue

On 2018-06-13 07:45 AM, D'Arcy Cain wrote: > On 2018-06-13 07:20 AM, James Cloos wrote: >>>>>>> D'Arcy Cain writes: >> >>>> Ie after both sides select t38, until they agree on the t38 terms. >> >>> OK, so does that mean that setting it

[asterisk-users] Database re-connect issue

I had a problem inserting CDR records into my PostgreSQL database. According to the log it failed to open the database at startup. I searched and found the following report. https://issues.asterisk.org/jira/browse/ASTERISK-15820 However, it looks like it was closed after 14 months and nothing

Re: [asterisk-users] T-38 re-invite issue

On 7/3/18 3:57 PM, D'Arcy Cain wrote: > On 2018-06-13 07:45 AM, D'Arcy Cain wrote: >> On 2018-06-13 07:20 AM, James Cloos wrote: >>>>>>>> D'Arcy Cain writes: >>> >>>>> Ie after both sides select t38, until they agree on the t38 terms. >&

Re: [asterisk-users] Block Spam Calls

On 12/12/19 5:33 PM, Greg Woods wrote: > Most spam calls are robocalls these days. At my house, I can block > pretty much all of the robocalls by requiring the caller to take some > action before ringing the phones. In our case, the action is just to > dial 1 for my wife or 2 for me. The only

Re: [asterisk-users] Faxes stopped working - AMI issue?

On 12/3/19 4:21 PM, Joshua C. Colp wrote: > You'd be getting more from AMI than just that. A message comes back when > you login, another when you actually do the originate, as well as events > most likely before then. The Asterisk testsuite uses AMI heavily, as do > others, so I'm confident that

Re: [asterisk-users] Faxes stopped working - AMI issue?

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 <551212> >     

Re: [asterisk-users] Faxes stopped working - AMI issue?

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

[asterisk-users] Splitting origination and termination

Up to now I have been using one remote server for both incoming and outgoing. The SIP entry looks like this: [combined] disallow=all allow=ulaw allow=gsm allow=ilbc dtmfmode=rfc2833 host=206.380.260.100 defaultuser=6477957868 secret=xx insecure=invite,port type=friend

[asterisk-users] Faxes stopped working - AMI issue?

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:

Re: [asterisk-users] Can't block intrusion

On 2020-04-02 08:01, Larry Moore wrote: > I suspect you have a good understanding of pf. Pretty good I think. As with everything I am always willing to learn more. > Have you included in your script running 'pfctl -k ' to kill > any states that may exists after you update your table? I

[asterisk-users] Can't block intrusion

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-4526] 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

Re: [asterisk-users] Can't block intrusion

On 2020-04-01 15:12, Greg Troxel wrote: > D'Arcy Cain 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 th

Re: [asterisk-users] Can't block intrusion

On 2020-04-01 16:28, Mark Boyce wrote: > On 1 Apr 2020, at 22:14, Greg Troxel > 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 from to any

Re: [asterisk-users] Can't block intrusion

On 2020-04-01 16:39, Larry Moore wrote: > Or the stateful entry still exists when the table entry is updated. > > Does your script also issue a command to kill existing states from that > host after it has updated the table, e.g.  pfctl -k 45.143.220.235 Yes, as I said in my OP that's the actual

Re: [asterisk-users] Can't block intrusion

> I haven't seen the issue today. One thing that I did was to move the > anti spoof line further up. I thought that anti-spoof would block > wherever it was. Could the location be important? Didn't matter. It happened again. I did do something though. I added a bunch of netblocks to my