Some background information.I have used Debian with Asterisk for several years. Have encountered zero problems. I am now trying to setup an Asterisk on a CentOS7 box using VMWare Workstation. I am brand new to CentOS and RHEL so I may be missing someth..
Author : dcropp
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Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS?Could anyone provide pros/cons for the various ASR options for Asterisk?We need the ability for very large grammars (over 100,000 options).Because of this, my initial thou..
I place a call into Asterisk (from SIP phone) and the To header does not have a tag.Asterisk then sends its Trying response, still no tag in the To header.The phone then replies with OK, this time the To header includes a tag.Is there any way to retri..
I need to make attended transfer work via an AMI request.Based on data from a Cisco trace from another system which successfully does an attended transfer, the Refer-To header requires the followi..
I to Originate channels using AMI.When the other end indicates the channel is ringing, I need to do some system notification work.Everything works great when the ITSP sends a 180 Ringing response.Through AMI events I see the channel state changed ..
I am working with a customer and their SIP provider is IPitimi.The customer needs to sometimes provide various caller id number for the calls going to IPitimi.They are processing calls for multiple businesses who want their caller id to show up.W..
We have a customer who does significant ConfBridge recording every day.They are concerned about the size of the recording that will accumulate.From the confbridge.conf.sample file, it mentions the default format is 8khz slinearIt is possible to cha..
I have a call coming in. I need to add a SIP Header to the channel. Then, I need to send the call to the Queue so it is sent to the Agent.The SIP header I added, I need to have appear in the INVITE sent to the Agent.It works in chan_sip.I send the c..
I am trying to set add a SIP Header to a call before adding it to the Queue.The dial plan sends the call to my macro to perform the work.When I use chan_sip, everything works as expected.When I use PSJIP support, its not adding the SIP header. Look..
I am running Asterisk 13.5.0.I have the Transfer working when using the chan_sip support. However, when I try to perform a Transfer using pjsip, it is failing.The one difference I am seeing in the SIP trace is chan_sip automatically sends the Referred-By.PJ..