Hi,
I'm trying to get Asterisk running with LDAP to be able to
authenticate sip user registrations. I'm using Asterisk
(1.8.13.1~dfsg1-3+deb7u3) on a Debian server.
Unfortunately I wasn't successful so far.
My res_ldap.conf looks like this (so pretty minimal):
---
[_general]
On 18-06-14 23:06, Linus Lüssing wrote:
Hi,
I'm trying to get Asterisk running with LDAP to be able to
authenticate sip user registrations. I'm using Asterisk
(1.8.13.1~dfsg1-3+deb7u3) on a Debian server.
Unfortunately I wasn't successful so far.
My res_ldap.conf looks like this (so pretty
Hi Patrick,
On Fri, Jun 20, 2014 at 02:46:06PM +0200, Patrick Laimbock wrote:
IIRC the recommendation in the latest Asterisk book is to use only a-z,
numerics (0-9) and underscore. So if you have [t...@chaotikum.org] in
sip.conf then that might not work because of the '@'.
I don't have the
On 20-06-14 15:05, Linus Lüssing wrote:
[snip]
having [test_phone_120d] in my sip.conf works fine. Ah wait - do
I need to have a user both in LDAP and sip.conf and the only
thing LDAP can do for me is the authentication/password checking?
As far as I know, yes :)
Cheers,
Patrick
--
Dear all,
Recently I switched from asterisk 10 (sip) to 12 (pjsip).
I converted sip.conf into pjsip.conf
In sip.conf there is:subscribecontext=blf_context
while in pjsip.conf i can't see that option any more.
How blf is handled by pjsip?
Best regards,
pepesz
--
pepesz wrote:
Dear all,
Recently I switched from asterisk 10 (sip) to 12 (pjsip).
I converted sip.conf into pjsip.conf
In sip.conf there is:subscribecontext=blf_context
while in pjsip.conf i can't see that option any more.
How blf is handled by pjsip?
Currently as there is no configuration
El 18/06/14 13:44, Alex Villacís Lasso escribió:
I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times,
What can I deduce from this? Is there some configuration on my
asterisk that can be tweaked so the failing requests can be handled
properly?
Is there additional information needed to make sense of this scenario
I suggest you capture the audio stream with tcpdump. You can then
convert the
Dear Joshua,
Thank you for your tip.
I put hints into context, however I still cannot make blf to work.
My small extensions and pjsip conf files are shown at
http://pastebin.com/fiwSHYd3
Asterisk 12.3.2
Phone Grandstream GxP2100 and 2010 with keys set as BusyLampField 6001 and
6002 respectively.
pepesz wrote:
Dear Joshua,
Thank you for your tip.
I put hints into context, however I still cannot make blf to work.
My small extensions and pjsip conf files are shown at
http://pastebin.com/fiwSHYd3
Asterisk 12.3.2
Phone Grandstream GxP2100 and 2010 with keys set as BusyLampField 6001
and
Dear Joshua,
Thanks for prompt response
It's clear now. Any clue/plans when/if it will be supported?
Best regards,
pepesz
On Sat, Jun 21, 2014 at 12:51 AM, Joshua Colp jc...@digium.com wrote:
pepesz wrote:
Dear Joshua,
Thank you for your tip.
I put hints into context, however I still
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