vicidial.org

This is a great application.... Thanks for your efforts.

I had a problem, where there was no sound when dialing out in auto mode. Sound was fine in manual mode. I have found other posts with same problem but no solution mentioned.

I upgraded to the latest Asterisk, 1.2.12.1 from 1.2.11. It now works!!!!

Running on SUSE 9.X. Using an X101p for timing. IAX link out to Binfone.


My next issue is to shorten the time taken to resolve sound on an auto call. It is about 6 seconds after answering the call. I have seen posts that suggest increasing the number of transfer statements in the extensions.conf. Which I will try....

exten => 8365,n,AGI(agi-VDADtransfer.agi,${EXTEN})

Asterisk 1.2.11 and 1.2.12 have a transcoding bug that breaks Local/ calls being redirected into meetme rooms. That bug is fixed in 1.2.12.1.

As for your sound resolving issue, Are you registering the account in iax.conf and then using the exact same registration string in your Dial string?

Yes the strings are same

I am doing testing by having the list call my cell phone.

I hear static on my cell phone after picking it up and then the static clears after 7 seconds. At this time communication can occur.

I am using IAX protocal, codec chosen is ulaw.

my cpu is only 300 mhz but my system is making only one call

300MHz CPU would do it. I can barely make one phonecall on my 500MHz test box.

I hate to beat a dead horse but feel that this is an important issue. I can find no information relating to the problem. I don't think it's a transcoding issue since ulaw is what is specified when making the call and ulaw is the final codec used.

I think this is a significant problem because it causes hangups. If there is any detectable delay for the callee they will and do hang-up.

My computer is only 300mhz but is making only one or 2 calls. It would seem that this would show up on faster computer when making 20 or more calls simultaneously.

I will do some manual mode testing. The problem should show up there as well but do not think that it does.

I will try upgrading the computer but have a feeling there is an issue here that needs to be resolved.

Yes, in manual mode the call seems to be resolved prior to dialing.

the dialing is done differently in manual mode. With Manual, the call is started from within the meetme conference so it never has to resolve.

I upgraded to the new version of vicidial, a 2.4 GHZ AMD, and have left the database on my old 300mhz computer.

I must commend you on an excellent upgrade!!!!!! The install went with no real problems. I had to fix the server_ip in my database, etc.

I saw no where though about setting up the cron user in the doc?

AND, most importantly the sound problem seems to resolve in < 2 secs.
Which will not be noticed by the callee.

Thank you for you excellent work!!!!!!!!!!!

I wish there were some way for me to help you in the development. Great job!!!!!!

Thank you for the compliments.

The user cron is no longer necessary as a Linux user in the 2.0.1 version.

As for supporting development, it always helps to buy the manuals from eflo.net or donate to the project

Any special consideration when installing VICIDIAL over Suse Linux?

allenrlevin wrote:This is a great application.... Thanks for your efforts.

I had a problem, where there was no sound when dialing out in auto mode. Sound was fine in manual mode. I have found other posts with same problem but no solution mentioned.

I upgraded to the latest Asterisk, 1.2.12.1 from 1.2.11. It now works!!!!

Running on SUSE 9.X. Using an X101p for timing. IAX link out to Binfone.


My next issue is to shorten the time taken to resolve sound on an auto call. It is about 6 seconds after answering the call. I have seen posts that suggest increasing the number of transfer statements in the extensions.conf. Which I will try....

exten => 8365,n,AGI(agi-VDADtransfer.agi,${EXTEN})

better try debian