On 06/15/2017 at 08:15 AM Michael Maier wrote:
> On 06/14/2017 at 10:17 PM, Joshua Colp wrote:
>> On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote:
>>
>>
>>
>>>
>>> I can now say, that asterisk / pjsip seams to work *mostly* as expected.
>>> Just one exception - and that's the package in
What might be good to consider is that "traditional" telcos actually use VOIP
tech on the back end a lot.
SIP URIs seem to be a kewl and ubiquitous solution, but for the average joe,
not so easy to setup and use.
And for the record, I consider wireless carriers as traditional telcos as well
Hey, so not particularly asterisk related, but I think the people on
this list are the right bunch to ask.
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?
I just setup an anonymous
Another "me too" (also Gmail).
I just received my 4th "account suspended, too many bounces" email,
after having several days of lost mailing list content over a short
vacation break the last time. When I notified the admin email account of
the failure, it seemed the responder missed the
On Thu, Jun 15, 2017 at 12:11:36PM +0200, Benoit Panizzon wrote:
> Or does anyone have an idea over what the asterisk is stumbling?
What if you set the maxdata in asterisk to a value lower than the other
side? e.g. sip.conf:
t38pt_udptl = yes,fec,maxdatagram=400
--
Hi all
I know, a fairly old asterisk installation.
Is there any way to debug T.38 handshaking issues?
We have a C3 Voice Switch with link to the Asterisk server.
I see this Dialogue:
C3 => Asterisk
=> Invite g711
<= 200OK
C3 detects Fax and send re-invite
=> Invite T.38
Version:0
On 06/14/2017 at 10:17 PM, Joshua Colp wrote:
> On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote:
>
>
>
>>
>> I can now say, that asterisk / pjsip seams to work *mostly* as expected.
>> Just one exception - and that's the package in question, which can't be
>> seen in tcpdump.
>>
>> I
