unable to barge calls in vicidial

[sayantandas]

Hi,

We have vicidial installed on ubuntu 10.10 x86 64bit server. details of the installation:

Vicidial: VERSION: 2.6-370a BUILD: 120529-2112 | Asterisk 1.4.39.1 | PHP5 v5.3.6 | Ubuntu 10.10 x86_64| Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Pentium(R) Dual-Core CPU E5700 @ 3.00GHz | 1 TB Hard Drive

We had installed this server from scratch following the instructions from vicidial wiki VICI:UbuntuInstall.

We have no issue in outgoing calls or recording. All works fine. Hoewver, if anyone wants to barge the calls, we cannot hear anything.

Also, the following is received at the terminal

Code: Select all

app_meetme.c: Unable to write frame to channel Local

when we click on BARGE

[williamconley]

Try an install from Vicibox 3.1.15 on a Virtual machine for testing. Try the same technique and see if it works. If it does ... then your installation has a bug. And since we do not have the entire install script (with any errors) nor the time to debug it ... you'll have to hunt through it or debug the barge to kill it.

That being said (why are you NOT using Vicibox?): Asterisk CLI from the barge attempt. Exact method hitting "barge" including the setup preceding the button push. It may also be necessary to use SIP debug and/or AGI debug in the asterisk cli to get enough information to find out what is going on, assuming it is not merely a configuration mistake (sending the call to a non-existent phone).

[sayantandas]

Try an install from Vicibox 3.1.15 on a Virtual machine for testing. Try the same technique and see if it works. If it does ... then your installation has a bug. And since we do not have the entire install script (with any errors) nor the time to debug it ... you'll have to hunt through it or debug the barge to kill it.

That being said (why are you NOT using Vicibox?): Asterisk CLI from the barge attempt. Exact method hitting "barge" including the setup preceding the button push. It may also be necessary to use SIP debug and/or AGI debug in the asterisk cli to get enough information to find out what is going on, assuming it is not merely a configuration mistake (sending the call to a non-existent phone).

Thanks. I have fixed the issue. Had to install libc6 i386 package.

[williamconley]

An interesting question would be how you determined that this was your fix? What command/error/log brought you to this conclusion? (Help out the next guy! Post your method! )

[sayantandas]

hi william, someone from this forum send me a pm with the solution. i tried as directed and it worked. i am assuming i was missing the 32 bit libraries.

[williamconley]

makes me wonder why they didn't post the answer here .. that's what this forum is for!