Re: [asterisk-users] Difference between Application Set and Function SET?

On Fri, Jun 16, 2017 at 12:41 PM, Jonathan H wrote: > It was only when I ran AsteriskLint over my dialplan that I noticed this: > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Set > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SET

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

I believe that Digium is using Mailman already (hence the in-the-clear monthly password reminders). I suggest that whoever administers the Mailman system should probably be able to tell why Gmail is bouncing (sometimes), and if not, there's plenty of active Mailman help available:

Re: [asterisk-users] Difference between Application Set and Function SET?

OK, thanks. That sort of makes sense. Is it case sensitive? Bonus quickie while I'm here (not worth own thread) - Asterisklint complains that: H_PAT_NON_CANONICAL: pattern '_#' is not in the canonical form '#' for the line exten => _#,1,Goto(s,1) I'm sure I read somewhere it should be _#. Am

Re: [asterisk-users] Difference between Application Set and Function SET?

> It was only when I ran AsteriskLint over my dialplan that I noticed this: > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Set > https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SET > > Hmmm, they both seem to do the same thing. Or don't they? In some

[asterisk-users] Difference between Application Set and Function SET?

It was only when I ran AsteriskLint over my dialplan that I noticed this: https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Set https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SET Hmmm, they both seem to do the same thing. Or don't they? Confused! --

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

I'm not sure of the precise specifics of how Digium runs the list, but this sort of problem has been a "known issue" with mailing list distributions ever since SPF and similar technologies showed up, almost a decade ago. DomainKeys and DMARC makes it more of an issue, but the overall problem is

Re: [asterisk-users] Difference between Application Set and Function SET?

On Fri, Jun 16, 2017 at 1:43 PM, Jonathan H wrote: > OK, thanks. That sort of makes sense. Is it case sensitive? > Is what case sensitive? Function names are case sensitive. Application names have historically been not case sensitive. > > Bonus quickie while I'm here

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

On Fri, June 16, 2017 12:28, Tim S wrote: Whether it is intentional or not these messages railing against the list operators has a decided tone of condescension which is not warranted. The fact of the matter is that DMARC is broken by design and the unpleasant effects that adoption of it has on

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

On Thursday 15 Jun 2017, Tim S wrote: > Whatever has been done, if anything, isn't working effectively. At this > point I'd like to see some response from the mailing list admin about any > root-cause efforts, AFAIC this is starting to smear the Digium/Asterisk > brand's ability to handle IT

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Fri, Jun 16, 2017, at 02:13 AM, Michael Maier wrote: > Has anybody any idea why asterisk drops the media stream in the 200 OK? > The channel has been T38_ENABLED before! Or is it necessary to add more > debug code? Who does the negotiating? > Only asterisk or is pjsip doing some parts, too?

Re: [asterisk-users] Is this the future of telephony?

On Friday 16 Jun 2017, Christopher van de Sande wrote: > So does anyone here think the traditional telephone company will go > extinct, and voice communication will take place via email like (or > equal to) sip uri's? Hardly! The job of the "traditional telephone company" has always been to

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

On Fri, Jun 16, 2017 at 08:38:59AM +0100, J Montoya or A J Stiles wrote: > > Whatever has been done, if anything, isn't working effectively. At this > > point I'd like to see some response from the mailing list admin about any > > root-cause efforts, AFAIC this is starting to smear the

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

On 16 June 2017 at 08:38, J Montoya or A J Stiles wrote: > It's hardly Digium's fault, if Google have decided that playing nicely with > syntactically-valid messages doesn't fit their business model Not really Gmail's fault, either. Someone above said they had

Re: [asterisk-users] Is this the future of telephony?

On Thu, Jun 15, 2017 at 08:56:29PM -0400, Christopher van de Sande wrote: > I just setup an anonymous endpoint in pjsip.conf and a context that > forwards to $EXTEN and when I setup the correct SRV records, it seems > that any SIP client that's smart enough can just dial my SIP/email > address.

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

Jonathan H wrote: On 16 June 2017 at 08:38, J Montoya or A J Stiles wrote: It's hardly Digium's fault, if Google have decided that playing nicely with syntactically-valid messages doesn't fit their business model Not really Gmail's fault, either. Someone

Re: [asterisk-users] pjsip: asterisk can't decide which codec to use

On 05/13/2017 at 07:21 AM Michael Maier wrote: > On 05/12/2017 at 08:49 PM, Joshua Colp wrote: >> On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote: >> >> >> >>> >>> If I'm doing exactly the same call originated with another extension, >>> there can't be seen these frequent changes. But the

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

Am 16.06.2017 um 11:12 schrieb Joshua Colp: On Fri, Jun 16, 2017, at 02:13 AM, Michael Maier wrote: Has anybody any idea why asterisk drops the media stream in the 200 OK? The channel has been T38_ENABLED before! Or is it necessary to add more debug code? Who does the negotiating? Only

Re: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'

On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote: > > t38modem and asterisk are using > > m=image 35622 udptl t38 >^ > > Provider uses > > m=image 35622 UDPTL t38 >^ > > Could this be a problem? If I'm sending internal only, it's always >

Re: [asterisk-users] pjsip: asterisk can't decide which codec to use

On Fri, Jun 16, 2017, at 12:29 PM, Michael Maier wrote: > I just tested your fix 2 times w/ using the scenario mentioned in the > bug report. It has been working for me. No more flipping. > > Asterisks indeed commits more than one codec in ok sdp, but always uses > the first one afterwards.

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

I'd hazard to say it probably is Digium's "fault", this was a recent and now consistent problem, which started within the last month or so. I'm on 7 other Linux-related mailing lists which all use similar mailer daemons, and none have this issue. I have been subscribed to Asterisk