Hello,
I've got a problem at the moment, that setting transmit_silence = yes
seems to have no effect on Asterisk 1.8-Certified.
Although it's enabled and core show settings confirms, that it is
really enabled, there are no RTP packets sent by Asterisk when waiting
for DMTF input or when Wait()
Hi John,
do you have a line in your sip.conf saying match_auth_username=yes ?
It needs to be in the general context or (I think) inside the peer
configuration.
I need to use this with user based auth, don't know if it's mandatory
for IP based auth also.
Greetings,
Max
signature.asc
Hi,
Maybe you have not allowed T.38 as acceptable codec ;-)
You can try with allow=all in your sip.conf.
Am 22.05.2013 16:39, schrieb Andrew Colin:
Hi guys,
Any idea why I am getting this error when someone tries to send me a T38
Fax?
--
Hi,
In 1.8 setting this flag it does not remove any caller id related data,
it just sets an information on how to handle the data.
If the direct dial target is a phone, it MAY show the callerid anyway,
but dialling out on a PSTN/VoIP trunk also sends the callerid.
But: If you place an emergency
Hi,
I'm trying to use AGC in combination with Asterisk 1.8 and an odd
telephone which is very loud when used with a headset and more quiet
when used normal.
Regarding to the documentation, AGC should be available since * 1.6 -
but every time I want to set it, the CLI tells me:
-- Executing
Hi Richard,
the macro you linked to did the trick for me - thank you!
Greetings from Wuppertal
Max Grobecker
Am 24.01.2013 00:18, schrieb Richard Mudgett:
- Original Message -
Hello out there,
I'm running an Asterisk 1.8.15-cert1 with DAHDI.
Today I noticed that Asterisk is
Hello out there,
I'm running an Asterisk 1.8.15-cert1 with DAHDI.
Today I noticed that Asterisk is signalling to the calling party the
current internal CallerID whenever I put a call to another internal phone.
Example:
Customer calls 020212345-555
- IVR answers and puts caller to the chosen
Hello,
I know about the german phone system that the sense of an anonymous call
is, that the called party has no way to get the caller's number in any way.
The last switch honours the anonymous bit and removes the phone
numbers before sending the call to the called party.
In EURO-ISDN you have a
Hi,
on a similar setup I set in sip.conf:
prematuremedia=no
progressinband=never
in the peers configuration.
With this config you tell Asterisk not to handle inband information at
all. But: Maybe you won't get any inband error messages also.
Greetings from Wuppertal
Max Grobecker
Am
09:45, schrieb Thorsten Göllner:
Did you take a look at
/var/log/syslog
/var/log/asterisk/messages
?
Using Debian? Take a look at iotop (apt-get install iotop). There you
can see information about which process consumes high io load.
Am 04.04.2011 17:23, schrieb Maximilian Grobecker
didn't get disturbed (no noises or crackles) but after about
five minutes Asterisk suddenly crashed without any further error messages.
Are you experiencing the same problem?
I'm really confused now why Asterisk crashes...
Thank you!
Maximilian Grobecker
Hello Thorsten,
the system has 4 GB RAM and about 2,5 GB free so swap space is not used
or exhausted.
Maybe the high load is not cause of this crashes but it's the only thing
the crashes can be reproduced with.
Thank you!
Maximilian Grobecker
Am 04.04.2011 16:03, schrieb Thorsten Göllner
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