Re: [asterisk-users] Dahdi get latest

2017-10-18 Thread Jean-Denis Girard
Le 18/10/2017 à 02:11, Jerry Geis a écrit :
> I am trying to use dahdi complete 2.11.1 with a 4.13 kernel. - NOT
> working for know reasons.
> I tried applying two patches but still get compile errors. AHHH!
> 
> How do I just use git to get the latest with the fixes 
> 
> This command did not work - I still get the errors.
> git clone git://git.asterisk.org/dahdi/linux
>  dahdi-linux

Hi Jerry,

Maybe you missed this patch:
https://issues.asterisk.org/jira/browse/DAHLIN-356

Or you can try my fork:
git clone -b next https://github.com/sysnux/dahdi-linux.git


Thanks,
-- 
Jean-Denis Girard

SysNux   Systèmes   Linux   en   Polynésie  française
https://www.sysnux.pf/   Tél: +689 40.50.10.40 / GSM: +689 87.797.527



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Re: [asterisk-users] user-agent access from pjsip

2017-10-18 Thread Bryant Zimmerman
I am trying to get the user-agent from extensions registered via pjsip. 
 With sip we could do a sip show peer peername and it would list the 
user-agent string. 
 In a pjsip deployment it looks like this info is likely in the contact. I 
know we can access it from the dialplan, but this is only works when a call 
occurs. How can we get the user-agent for extensions from the console. We 
need this for firmware version checking of extensions as many providers 
include that in the user-agent.  Any ideas as the pjsip show contact 
contactname does not return any real helpful info to the command line. 
  
 Please advise if you are able. 
  
 Thanks
 Bryant

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[asterisk-users] PJSIP Asteirks 13 - Audio Jitter in one direction only

2017-10-18 Thread Bryant Zimmerman
?We have upgraded a system from Asterisk 11 to Asterisk 13 with pjsip.
 We are experiencing random Jitter on outbound calls. This was not occurring 
when running asterisk 11.

 We have two IP's bound to pjsip one on the private vlan network the phones are 
on and the asterisk one on the asterisk wan vlan. We record the calls on the 
asterisk switch so we have the call legs. It appears that the audio is making 
it to the switch fine, but is being garbled before it leaves asterisk to the 
destination carrier. We have all media running through the server and this is 
happening when there is only 1 to 2 calls on the line. The cpu, and memory are 
not even being pushed. We are running G711 as the codec so there should be no 
real transcoding occurring..

 What could be causing this. The users are very upset. This is a very transient 
issue so the breakup is can occur for two to four seconds and then goes away. 
It is like asterisk and pjsip are screwing with the audio. Please advise.

 zktech

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[asterisk-users] Dahdi get latest

2017-10-18 Thread Jerry Geis
I am trying to use dahdi complete 2.11.1 with a 4.13 kernel. - NOT working
for know reasons.
I tried applying two patches but still get compile errors. AHHH!

How do I just use git to get the latest with the fixes 

This command did not work - I still get the errors.
git clone git://git.asterisk.org/dahdi/linux dahdi-linux

Thanks,

Jerry
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