I think the WARNING on Gosub() calls from AEL could change to NOTICE; or
at least get more specific, such as "instead of calling Gosub()
directly, please try an AEL macro."
A new AEL keyword to eventually replace "macro", such as "subcontext",
might help clarify things, too.
But it should work
On 2019-11-26 09:17, Greg Troxel wrote:
> For the second issue, it would be nice if Dial just discarded empty
> destinations, as in
>
> Dial(PJSIP/foo&)
> Dial(PJSIP/foo&/baz)
>
> as would result from the following if there were no bar registrations
>
>
On 2019-11-26 20:05, Greg Troxel wrote:
> "C.Maj" writes:
>
>> Another option for a patch would be to extend the PJSIP_DIAL_CONTACTS
>> function with an argument such as 'please' to minimally return the
>> endpoint name in a Dialable format when no reachable co
On 2019-12-19 06:10, Antony Stone wrote:
> On Thursday 19 December 2019 at 14:04:36, Jerry Geis wrote:
>
>> I presume it would just be sending a SIP message - no need to get anything
>> back. Just want to pop a message on the phone.
I think there are some Polycoms that support RFC 3428 SIP
/
* *
* Always Be Conferencing (ABC)*
*
On 2020-01-15 08:33, David P wrote:
> We use Asterisk 14 to proxy calls between two servers, 10.0.0.192 to
Asterisk 17 is the most recent major version, but I don't know if an
upgrade will solve your problem.
> 10.0.0.228. But sometimes another of our servers becomes listed as a SIP
> agent,
On 2020-01-15 11:24, Administrator wrote:
8<'s
> One of the provider took a pcap and told us that expiration was set to 0
> that's why they don't accept the registration. We took a pcap on our
> side when SIP packet goes out of our server and we see that the
> expiration parameter is setted to
On 2020-01-16 02:16, Administrator wrote:
>
> Le 15/01/2020 à 19:50, C.Maj a écrit :
>> On 2020-01-15 11:24, Administrator wrote:
>>
>> 8<'s
>>
>>> One of the provider took a pcap and told us that expiration was set to 0
>>> that's why they d
Howdy y'all,
Penguin PBX Solutions is pleased to announce the release of Always Be
Conferencing version 16l, released under the Creative Commons Zero v1.0
Universal (CC0 1.0) license for maximum efficiency in distribution and
adaptation into your local environment. Just one AEL file & full of
On 2020-08-18 13:00, Carlos Chavez wrote:
> users complain that confbridge calls end after about 30 minutes or so
You might want to turn up SIP debug logging -- could be a re-INVITE is getting
dropped, NAT pin-hole is closing, or some other network issue.
--
鸞 C. Maj, Technology Captain @
On 2020-05-28 10:15, Doug Lytle wrote:
> Everybody,
>
> I've had a request from my manager that I figure out how to get our Asterisk
> 13.x system using chan_sip to be able to display on the Polycom VVX series
> phone display (firmware 5.9.5), when an extension is called and the person on
>
On 2020-05-28 11:10, Doug Lytle wrote:
But if you've already got the caller on the phone, then you might consider
the CONNECTEDLINE function in Asterisk...
>
> And that we don't.
>
> It's the third party that would like the notification the the destination
> phone is currently busy
On 2020-12-09 12:54, Dovid Bender wrote:
> For older versions of Astrerisk (1.8, 11, 13) what is everyones
> experience when it comes to core count vs higher speed cpu's? We seem to be
> hitting some cpu issues. In the past we went for more cores but now I
> wonder if we should go with fewer cores
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