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Asterisk CDR On Call Transfer

I have searched for some time but I have not found an answer on how to fix the CDR when a call is transferred. The problem is that if someone dials a cell phone and then transfers the call to another extension the CDR for the cell call stops and there is no way to track that the call was transferred so we can bill correctly. Many people have asked this question but there is no answer, only a mention that it should be fixed in 1.6 which it is not (at least on 1.6.2.11).

Any tips oh how to correct this problem? A lot of customers give me grief about this because they cannot properly bill people for their cell calls.

Asterisk AGISIGHUP

I am setting AGISIGHUP = no before running a Perl script via AGI but it doesn’t seem to be doing anything as the script is still exiting on a hangup and not completing properly. I am using Asterisk 1.4.35 and have tried various combinations. Can anyone shed any light on this?

Regards
Lee

Asterisk ACK BYE question

We’re running Asterisk 1.6.1.17 for our campus voicemail server and Juniper M120s as our SBC. Unanswered calls, which arrive via the SBC, are diverted to voicemail using a 302 redirect when the called party doesn’t answer. In this case the caller is able to hear the greetings and begin to leave a message only to have Asterisk terminate the call mid-recording.

We’re uncertain why this is happening and this is where we are hoping you can help. In our environment the caller is any set on the PSTN. They call one of our IP phones which no one answers. Our proxy, SER, responds to the SBC with a 302 redirect and the call is diverted to Asterisk. The caller hears the unavailable greeting for 6-4050, begins to leave a message and is cut-off after about 10 seconds. In an ngrep trace we see Asterisk receive an ACK from the SBC and it immediately responds with a BYE message for that call.

Has anyone else experienced this type of issue?

Asterisk Problem With One Way Audio With No NAT

I have a soft phone (VoIP Phone)– expresstalk– on a computer in my network and I use the internal ip address of the Asterisk box to register the phone. But using asterisk-1.8 between revisions 281912 and 281982 it breaks — after a few seconds of the call, I lose audio from the asterisk box to my soft phone, but not the other way around. This looks like one commit, but obviously I would like to know what’s going on here?

Thanks in advance for any ideas.

Asterisk GUI in Version 1.6

The ISO download appears to have Astersik 1.6 with the FreePBX GUI but I am looking to use the Asterisk GUI. The only option for the Asterisk GUI is to use 1.4. Is it as simple as installing 1.6 only then using the yum repository to install the Asterisk GUI? If so, what packages are needed?

Thanks!