We recently had a major crash on our Asterisk phone system and the following error was found in the log.
Jul 19 13:51:50 WARNING: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/3 already in use on span 8. Hanging up owner.
Unable to determine if this error was catastrophic enough to hang up all callers but this is all we have to work with right now.
I have heard through the grapevine that Asterisk does not play nice when an agent tries to transfer a call to another agent who is already logged in. Does the error above resemble that scenario?
I also have on good authority that asterisk 1.2.10 has a fix for this bug? Can anyone comment on that as well.
Any assistance is appreciated?
i’m not an expert with this side of things, but it looks like there was a ‘collision’ of sorts, in that a Zap channel was in use and Asterisk attempted to grab it.
this might be a problem in your dialplan, especially if you have specific zap channels being called for outbound dialing (like Dial(Zap/1/15555555555)…)
did the whole system crash or did you just lose all your calls? be a bit more specific and we can probably get this sorted out.
Agents were not receiving new calls or able to login/logout (sounds like Asterisk crashed to me) but not certain if they were able to finish the call they were on. I will confirm.
This call center has 99.9% of the traffic as inbound calls. Outbound calls are seldom made but I can get more specifics there as well. We don’t have any specific rules on using certain zap channels for any outbound calls but I will locate the dial plan and get back to you very soon.
While I try to gather the requested info I would also like to offer up some additional info.
Collisions on the zap channels is somethine we beleive are occuring as well. We are using Citel Gateways with Nortel digital phones on this installation and these gateways (four of them) are connected to Asterisk server via an 8 port switch (made by 3com) We believe this 8 port switch is not robust enough to handle all the traffic we are delivering through to the Asterisk server. A new more powerful 24 port managed switch is being shipped out today as a temporary fix.
The more permanent plan is to move away from the Nortel phones and Citel gateways completely. We have been testing the Aastra 480i sip phone with fantastic results so far (third day on two phones) and looks like these will be the replacement for the Nortel phones and Citel gateways within the next two to three weeks.