Re: [asterisk-users] PRI error: link goes down when make calls

2016-09-22 Thread Mc GRATH Ricardo 
Hi.
These becomes from layer 1 issue and could becomes from a wide of  
possibilities (telco or Asterisk PRI card).
At first check cables an connectors, on Asterisk could make a loopback and 
check if problem persist.
On telco should check cable of DSL Modem or whatever provide PRI service.

Mc GRATH Ricardo
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[asterisk-users] Asterisk 14.0.0-rc2 Now Available

2016-09-22 Thread Asterisk Development Team 
The Asterisk Development Team has announced the second release candidate of
Asterisk 14.0.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.0.0-rc2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release candidate:

Bugs fixed in this release:
---
 * ASTERISK-26397 - manager: PresenceState action crashes Asterisk
  14 (Reported by Andrew Nagy)
 * ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by
  Joshua Colp)
 * ASTERISK-26391 - Consoles do not display verbose logger messages
  even when requested. (Reported by Marcelo Terres)
 * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled
  (Reported by Anthony Messina)
 * ASTERISK-26365 - rtp: Offer with multiple payloads for same
  codec is incorrectly handled (Reported by Joshua Colp)
 * ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten
  for connectionful protocols (Reported by Joshua Colp)

For a full list of changes in this release candidate, please see the
ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.0-rc2

Thank you for your continued support of Asterisk!
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[asterisk-users] Planned maintenance for community services Friday evening, Sept 23rd, 2016

2016-09-22 Thread Digium's Asterisk Development Team 
Tomorrow evening, community services may have intermittent
availability due to maintenance. This maintenance will begin at
approximately 6:00 PM CDT[1] and is expected to last only a few
minutes and no longer than one(1) hour, ending around 7:00 PM.

The affected services are:

* All Asterisk community services including everything in the
asterisk.org domain.

Thank you for your support!

[1]: http://bit.ly/2cOL3kG (see converted times)

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[asterisk-users] PRI error: link goes down when make calls

2016-09-22 Thread Rafael dos Santos Saraiva 
Hi

I have a PRI link with a Brazilian telco when i make a call from Asterisk
to PRI link the call doesn't complete and the link goes down (RED alarm),
after this returns to status OK. Incoming calls works, but whitout audio

The following link has the log on call at the moment when the link becomes
down:

http://pastebin.com/V5ySVc0f

Any suggestion?

Thank's


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[image: Sua Foto]  Rafael S. Saraiva
Porto Alegre - RS | Mobile:  (51) 8174-7956


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Re: [asterisk-users] Trouble getting peer variable (sip username) on 302 Moved Temporarily

2016-09-22 Thread Jonas Kellens 

On 02-09-16 11:51, Administrator TOOTAI wrote:

Le 02/09/2016 à 11:26, Jonas Kellens a écrit :

Hello

when setting a local forward (in this case to extension 23) on a SIP
phone, I see the following on the Asterisk CLI :


[Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back
from 11.22.33.44:40670
[Aug 31 14:59:34] -- Now forwarding
Local/myaccount184@CallFromQueue-07f4;2 to 'Local/23@from-internal'
(thanks to SIP/myaccount184-3729)


Question : how can I read the variable which contains the value
'myaccount184' in the context from-internal ?


From SIP_HEADER(TO) ?

[...]



Hello


SIP_HEADER(TO) is empty. So is SIP_HEADER(FROM).


My dialplan :

exten => _ZXX,n,NoOp(CallerIDnum = ${CALLERID(num)} CallerIDall = 
${CALLERID(all)})

exten => _ZXX,n,NoOp(sipheaderto = ${SIP_HEADER(TO)})
exten => _ZXX,n,NoOp(sipheaderfrom = ${SIP_HEADER(FROM)})


On the Asterisk CLI :


[Sep 22 09:43:04] -- Called SIP/nnsa135
[Sep 22 09:43:04] -- Got SIP response 302 "Moved Temporarily" back 
from 8.9.10.11:65466
[Sep 22 09:43:04] -- Now forwarding SIP/Incoming-0bd9 to 
'Local/208@from-context' (thanks to SIP/nnsa135-0be1)

...
[Sep 22 09:43:04] -- Executing [208@from-context:5] 
NoOp("Local/208@from-context-0079;2", "CallerIDnum = 09210 
CallerIDall = "Cpss" <09210>") in new stack
[Sep 22 09:43:04] -- Executing [208@from-context:6] 
NoOp("Local/208@from-context-0079;2", "sipheaderto = ") in new stack
[Sep 22 09:43:04] -- Executing [208@from-context:7] 
NoOp("Local/208@from-context-0079;2", "sipheaderfrom = ") in new stack





Any more ideas on how to get the value "nnsa135" (being the SIP 
username) please ?





Kind regards.


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