hello every body
i have problem in receiving fax from e1 lines. this is my scenario:
faxphoneericson pbx ---e1asterisksip-zoiper-softphone
when i send fax from zoiper, i can receive it successfully on the faxphone
but when i send fax from faxphone, i can not receive it on zoiper.
Hi!
I am running an Asterisk PBX 11.6-cert10 with about 20 SIP phones and recently one of the phones
(Snom 720) always returns "486 Busy Here" when calling anonymously. It's only a single phone,
the rest works as expected. I checked the phone's settings and there are no differences in the
conf
Hi,
every two weeks the asterisk process has a segfault. Any idea whats reason or
what I can do...
thanks
pc kernel: [1780743.239296] asterisk[11362]: segfault at 0 ip (null)
sp 7f1e396b04a8 error 14
version is debian jessie
Asterisk 11.13.1~dfsg-2+b1 built by buildd @ brahms on a
You'll want to follow these instructions to get a backtrace:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
And then create an issue here and attach the backtrace file:
https://issues.asterisk.org
This way the Asterisk team will have the best chance of being able to
locate and re
Hi,
I have a queue configured on Asterisk (11.14.2);
Currently regardless of what codec any side is using,
every call handled by the queue is seen as 2 separate calls (4 channels).
Calls are incoming to the queue from a (few) SIP trunk(s),
queue has the 'queue no answer' option set, autofill is O
I am running an Asterisk PBX 11.6-cert10 with about 20 SIP phones and recently one of the
phones (Snom 720) always returns "486 Busy Here" when calling anonymously. It's only a single
phone, the rest works as expected. I checked the phone's settings and there are no differences
in the config
Hi list,
I've been googling this issue and found some good resources however I am still
running into problems with the following combo ... Here's my story;
- Asterisk 13.4 with FreePBX 12.
- Migrating from Asterisk 11 / FreePBX 2.11
- Mix of Cisco 79xx handsets, mostly 7940G's.
I had exact same issue with pjsip instead of sip - I was able to solve it
by setting the password to blank. But I switched to asterisk 11 because the
chan_mobile module was giving me troubles in 13.
On Wed, Jul 22, 2015 at 7:07 AM, Brendan Ord
wrote:
> Hi list,
>
>
>
> I’ve been googling this
I’ve gotten to the bottom of this;
Seems that the pjsip.endpoint_custom.conf isn’t getting included properly, or
my syntax is wrong.
If I put force_rport=no into pjsip.endpoint.conf and reload only Asterisk,
everything works perfectly. Unfortunately, I’m using FreePBX, so it owns this
file an
