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GoToIfTime() Occasionally Skips Valid Rule

My extensions.conf contains the following rules to control our IVR and make sure after hours calls are sent to the appropriate extension.Code: same => n,Answer(); this sets the inter-digit timersame...

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GoToIfTime() Occasionally Skips Valid Rule

First step, are you sure that your box's time is correct? If you're using NTP, is it synchronizing against a reliable source? Is your timezone set correctly?You can also add some logging to your...

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Asterisk 13 PJSIP : "Contact" field in SIP is randomized

Hello,We are using Asterisk 13.1-cert2 and PJSIP, with Nextiva as the SIP Trunk provider.Incoming calls are working, but dialing out is not working. The carrier says that there are 2 reasons why...

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[BUG?] Asterisk crashes randomly, possible ODBC issue?

Hello all,I've got a weird issue on a new Asterisk 13.6.0 install. This is a fairly complex install, with ODBC logging for CDR and queue_log, and heavy use of multiple queues and AGI applications....

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[BUG?] Asterisk crashes randomly, possible ODBC issue?

In the hangup context 'h', have you tried DeadAGI instead of AGI ?Also, what happens when you do not load the ODBC modules ?Statistics : Posted by csyria • on Mon Jan 04, 2016 12:20 pm • Replies 3 •...

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[BUG?] Asterisk crashes randomly, possible ODBC issue?

DeadAGI is obsolete.Statistics : Posted by csyria • on Mon Jan 04, 2016 12:20 pm • Replies 3 • Views 1713

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[BUG?] Asterisk crashes randomly, possible ODBC issue?

penguinpbx wrote:Also, what happens when you do not load the ODBC modules ?This is a prod box so not having CDRs isn't going to fly... and unfortunately, it only seems to happen rarely, so I need those...

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Unable to create channel dahdi

Here is my configuration for an analog diguim 8 port card in the system.conf file,Code: loadzone     = usdefaultzone  = usfxsks = 1,2,3,4,5,6,7,8echocanceller = mg2,1-8Below is the dahdi.conf file...

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Unable to create channel dahdi

You haven't defined group 4.Statistics : Posted by tella • on Mon Feb 01, 2016 2:05 pm • Replies 2 • Views 72

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Unable to create channel dahdi

Yes - david55, I will try to define the chanel 4.I am using dahdi version 2.10.0.1. While asterisk and libpri using the current version, do you think is that a problem?Look forward to your reply.Thanks...

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SOFTSWITCH TO ASTERISK CALL - SIP SETTINGS

HelloI have questionMy friend using MVTS(softswitch) for voip.He want send the traffic to usİ havent switch i have only one server and i have elastix But i cant configure SIP trunk incoming settings...

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SOFTSWITCH TO ASTERISK CALL - SIP SETTINGS

You can probably do it.The incoming trunk should be matched by IP address, like:Code: host=1.2.3.4Please consider adjusting your firewall to only accept traffic from known hosts.Also you might try...

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Asterisk MixMonitor Recording FileName

Hi allWe have an Asterisk Server (1.8.10 version) with more than 4 Queues. We need to manage files of Recorded calls on incoming and outgoing conversations.When an incoming call transferred multiple...

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Asterisk MixMonitor Recording FileName

You might consider the Asterisk Channel Event Logger (CEL) interface for logging of step-by-step actions.Changing the file name is a separate issue, although tracking the UniqueID is a good way to do...

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newie

g'dayi am newie with asterisk and need to get up and running a small office for a mate comprising of around 6 Avaya 4610 Sw Ip supporting POE handsetsappreciate a prompt assistancea) minimum specs for...

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newie

Quote:a) minimum specs for a server to run asterisk latest version, mate will buy a 2nd hand machineA router running OpenWRT could do it. But a gently used floor model PC will probably compile a little...

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Problem with IVR Barge-In for PSTN calls

Hi! My configuration is Asterisk 11 + OpenVox D230E E1 card + UniMRCP + Speech recognition server. I implemented voice barge-in during IVR playng promt ( it stops and start recognize ). Everything...

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non-critical invite transaction.

Fix the network.Statistics : Posted by geraldinejns • on Sat Jan 30, 2016 11:20 am • Replies 5 • Views 240

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non-critical invite transaction.

the voip phone rings but does not connect to the auto attendant (ivr )Statistics : Posted by geraldinejns • on Sat Jan 30, 2016 11:20 am • Replies 5 • Views 240

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