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Having an issue with manual dial campaigns, everything works except sporadically the hangup button doesn't actually hang the call up. It will pop up the disposition screen. However, if the call didn't disconnect and if the agent where to hit dial next number, they will have two calls stacked in their conference. Any ideas/thoughts/know issues or fixes?

There are some issues with the channel name actually changing between ring-time and Answer. I have noticed this especially on SIP, not so much on Zap or IAX.

So when you say SIP, are talking about SIP Phone or SIP at the carrier level? Some of my routes are IAX2 and some are SIP...I have SIP phones as well

SIP carrier, but that was with a much earlier version of Asterisk, I really don't have any client who are currently doing a lot of manual dialing through SIP trunks so I'm not sure if it is still the case.

hi

does this have any chance that congestion() is being used rather than hangup() in the extensions.conf, I read an earlier thread about a similar issue?

regards

devafree

Nope, congestion is no where used in my extensions.conf

However, I am using 1.2.12 as I got that version from internal "loads" server and thought that was the newest flav of the 1.2 version....I know it isn't. Matt, when you say earlier version, do you mean earlier than 1.2?

There's a problem with 1.2.10, 1.2.11 and 1.2.12. You should upgrade to at least 1.2.12.1. There was a major bug in the codec selection process that messes up all sorts of things.

hmmmm....actually I am on version 1.2.12.1...Also am getting this issue now with load balancing when utilizing more than one server and multiple agents...Thanks Matt

What is the loadavg on the server when this happens?

Load balancing with 2 servers, 1st (trunks set to 5) and 2nd (trunks set to 96). 10 agents logged in dialing about 20 calls:

Server1: load average: 0.13, 0.14, 0.09 Idle never below 87%
Server2: load average: 0.05, 0.02, 0.00 Idle never below 95%

Within 2-3 minutes it will start so I have to undo the load balancing...using extension 8367 or 8368 does the same thing.

On Second and Third (When trying with 3 servers) will get this error from time to time:

-- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 66.235.230.10

Thanks,

-Chad

I'm not really sure what would cause that error, but have you considered upgrading to a newer version of Asterisk?

Hi

Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 66.235.230.10

this could be cause of Codec translation issue

as per some asterisk forums.


Ram

Thanks ramindia...Yes, I have been reading a little bit about the sip response and have seen mentions of reinvite and or nat settings... Matt, yes I have considered upgrading as well...this issue is more towards my multi-server setup posting and will continue with that issue first as I'm not doing any manual dial campaigns anymore at this time.

hi

Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from 66.235.230.10

If your getting this only occasionally, it could be becoz of mailbox='xx' setting in sip.conf, where the sip client is like thru an audiocodes or some ATA that doesnt support voicemail box alerts.

regards

devafree