On Tue, Oct 27, 2020 at 3:25 PM Olivier wrote:
> Thanks Joshua for replying !
>
> What would you advise :
> - leaving STUN address empty, in rtp.conf, as "STUN is not required for
> ICE"
> - configure it with one public STUN (I'm using stun.voip.ovh.net for this
> but I don't know how this
Thanks Joshua for replying !
What would you advise :
- leaving STUN address empty, in rtp.conf, as "STUN is not required for ICE"
- configure it with one public STUN (I'm using stun.voip.ovh.net for this
but I don't know how this server really works)
Cheers
Le mar. 27 oct. 2020 à 09:53, Joshua
Jon,
We are only using FastAgi. On the second system (running Asterisk 16) there
are no agi's running (just some bash scripts on call hangup). I did add
some hackey code (netstat -nua | grep -v 'udp0 0' | grep -v
udp6 | grep -v ' 0 0.0.0.0' | grep udp) to my bash script to check out
El Tue, 27 Oct 2020 12:52:47 -0400
Dovid Bender escribió:
> Hi,
>
> Sorry in advance that I am emailing the users list and not the biz list I
> think I will find my target audience here. We are looking to hire a
> consultant to help us figure out an issue. We are having what seems are
> "random
Hi,
Sorry in advance that I am emailing the users list and not the biz list I
think I will find my target audience here. We are looking to hire a
consultant to help us figure out an issue. We are having what seems are
"random load" issues with bare metal boxes that are dedicated to Asterisk
and a
Greetings all,
This is a reminder that as of October 24th Asterisk 13 has gone security
fix only. This means that it will receive no new bug fixes, and any
outstanding bug fixes which have been merged (or are up for review) will be
released in an upcoming final bug fix release. Asterisk 13 has
Hi.
I've discovered a bug in the Dial() string processing (for Asterisk 13.14.1 at
least).
According to the documentation in channels/chan_sip.c the Dial() string syntax
is:
* SIP/devicename
* or SIP/username@domain (SIP uri)
* or
On Tue, Oct 27, 2020 at 5:35 AM Olivier wrote:
> Hello,
>
> Where can I find doc about PJSIP's ice_support parameter ?
>
> Do you need to configure things elsewhere in Asterisk config files
> (rtp.conf, PJSIP transport sections, ...) to make ICE work properly ?
>
It needs to also be enabled in
Hello,
Where can I find doc about PJSIP's ice_support parameter ?
Do you need to configure things elsewhere in Asterisk config files
(rtp.conf, PJSIP transport sections, ...) to make ICE work properly ?
I'm asking because, if I'm not mistaken, STUN requires setting a STUN
server so I think ICE
Hello,
Is it possible to set different features.conf dialing sequences (atxfer,
pickup, ...) for different users ?
For instance, what if I want Alice to dial *8 to pickup a call and Bob to
dial ** to pickup calls ?
I can see that features.conf includes application maps but can these be
used for
Hello,
In project, a customer has two WAN access. More precisely:
Internet - --- Router1 --- FortiGate Firewall Router
-- Asterisk
| |
- --- Router2 --
Both WAN
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