ens or even hundreds of them).
Regards and thanks,
Recursive
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lems since I restarted
Asterisk to make it work again (yesterday in the evening).
Thank you very much for any answer, hint and bit of background information.
Regards,
Recursive
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On 09.11.2015 15:54, Olivier wrote:
>
>
> 2015-11-09 15:06 GMT+01:00 Recursive <mailto:li...@binarus.de>>:
>
> On 06.11.2015 14:06, Rusty Newton wrote:
> >
> > It should be able to handle typical regular expression. I don't see
> >
e way. That seemed logic to me and lead me into thinking that
the pattern string in the REGEX call could be affected by this; hence my
initial post.
I am apologizing that I haven't found the bug in my dialplan at an
On 06.11.2015 14:06, Rusty Newton wrote:
> On Thu, Nov 5, 2015 at 12:49 AM, Recursive wrote:
>>
>> same => n(A1), GotoIf($[${REGEX("^\+49.*" ${EXTEN})}]?:A2)
>> [..]
>> Actually, the REGEX function is not able to handle normal regular
>> express
ndle normal regular expressions.
To make things worse, there doesn't seem to be any documentation. Could anybody
please point me to documentation or tell me how write that very simple pattern?
Thank you very much,
Recursive
P.S. This happens in Asterisk 13.6.0 - I haven't tested with
I am quite
sure that this affects every Asterisk / PJSIP installation ...
Thank you very much,
Recursive
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On 14.10.2015 21:59, Eric Cooper wrote:
> On Tue, Oct 13, 2015 at 06:54:51PM +0200, Recursive wrote:
>> Some of the software components involved is inserting an unwanted
>> header line into the fax. The header is always formatted the same
>> way; an example (using a two page
and there are German locales installed on the
Linux system where Asterisk runs on, but the system-wide default locale on that
system is en_US.UTF-8 (i.e. LANG=en_US.UTF-8).
This is on Debian Wheezy, Asterisk 13.5.0 and PJSIP 2.4.5.
What do you think about that situation?
Thank you very much,
R
ble
the core dumps like shown in [1].
Regards,
Recursive
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respective part of the source code. I think I am not (yet ?) so deep in
Asterisk that I could help debug the problem, but if anybody tells me what to
do I am willing to help.
I hope that the attachment won't get blocked ...
Regards,
Recursive
*CLI> -- Removed contact 'sip:hfc1z...@
ching extension (pattern) in
the dialplan? Is there some hook to manipulate the causecode?
There seem to be some dialplan applications which are sending codes for "busy"
or "congestion" to the other par
On 06.01.2015 13:44, Frederic Van Espen wrote:
> On Thu, Jan 1, 2015 at 7:09 PM, Recursive wrote:
>> 1) Did anybody test T.38 with SPANDSP? If yes, which version of SPANDSP did
>> you use? Mine is 0.0.6 PRE 20. Should I try to upgrade to PRE 21? Or to one
>> of the snaps
he log and configuration
available for download (can't provide them here due to the 40 kB message size
limit).
Thank you very much for any thoughts,
Recursive
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e somebody with appropriate knowledge is willing to take a
look into the log and to tell me in simple words if these lines denote a normal
situation or if they denote a problem.
Thank you very much,
Recursive
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f the RFC.
Thanks to the hints you have given, if I find out how to configure the media
address, I think I am able to change the owner line. Furthermore, I'll try to
let Asterisk bind to all IP addresses or to the main one; that way, the owner
address would still be a local one,
Thank you very much for any ideas,
Recursive
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res_pjsip / chan_pjsip
instead.
Thank you very much again to all who tried to help.
Regards,
Recursive
On 02.12.2014 09:24, Recursive wrote:
> Dear all,
>
> I have the following situation:
>
> Local T.38 endpoint <-> ASTERISK <-> SIP provider (with T.38 support)
>
&g
.
So the problems seem to arise only when Asterisk is in the middle.
I did not test the receive direction yet.
I'll now backup my Asterisk configuration for future reference and research and
start to try all the suggestions from the helpful people here step by step.
I'll report bac
x (regardless of which configuration
parameters I was using).
Matt, I nearly don't dare to ask, but could you eventually take a quick look
into the logs I have provided? Do you see any reason why asterisk hangs up,
claiming a critical packet timeout, although all packets seemingly have bee
27;m almost there, it is not important for me who invites whom, but
I think the key to success would be to find out why Asterisk hangs up the call
prematurely.
Of course, I will try all your suggestions. But I am also still seeing a chance
to solve the problem by look
ou could refer to my message from some minutes ago. I
have provided all the details there.
Thank you very much,
Recursive
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as contact. But in the
original invite in packet 28879, the contact was
0049...@spock-asterisk.home.omeganet.de. Is this bad?
Thank you very much in advance to everybody who tries to help.
Regards,
Recursive
This is my sip.conf:
[general]
sessi
always makes the SIP provider
switch to T38 at once at the very beginning of the transmission which of course
won't work. I think I'll open another thread about that problem; it's somehow
off-topic here.
Regards and thank you very much,
Recursive
On 10.12.2014 11:42, Frede
(as all parties already have agreed
>upon T.38).
Thus, is my endpoint really misbehaving, and if yes, is there anything I can do
about it on Asterisk's side? Or do the SIP/T.38 state machines allow such
(seemingly superfluous) re-invite, and it's Asterisk's fault to an
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