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rrb555 wrote:probably with your voip? is this happening before? do a traceroute using mtr to your voip server ip.
williamconley wrote:1) Your Vicidial version is very old. Seriously consider upgrading.
2) You listed your vicidial version but not your installation method with version (and this question could be installation related)
3) You did not mention if this is a fresh install or recently reactivated system that has been dormant for a few months/years. Probably a good idea to mention if this problem "just popped up" on a server that has been in daily use for a few years AND mention any recent changes (like ... new carrier last week?).
4) Have you tried this with a second carrier? Obviously if all the calls that experience this issue are on a single carrier and none are on the 2nd carrier, the carrier would be the problem ... but that test only works if you have two carriers. LOL
5) If you do not have three carriers yet, I'd say you just found your reason to get three carriers.
6) All agents? Some agents? have you begun to log these events yet to see if there is a common thread among them? If not, this would be a good time to start. Get an email address at your organization that agents can send their trouble report to, or put it in a DB or a log book or a wordpress post page ... something. Include call date/time/lead_id/agent: user & phone.
williamconley wrote:Time for a second carrier. Overdue, in fact, if you've been running with One carrier for this long I'm surprised you haven't had a problem before now.
williamconley wrote:Not if the call has not dropped but merely lost audio from the carrier. There is the possibility that the call (from asterisks's point of view) has not changed. Just ... silence.
But if you had a second carrier, you'd have dispensed with this possibility already ... or determined that the carrier is the problem.
I guess you could argue with deep logs and fun testing for a few days first, though. Like trying to rewire the car because it didn't start (since you don't want to purchase a gas can). Of course there May be another problem ... but trying to put gas in it is good place to start. LOL
williamconley wrote:iftop would undoubtedly stop showing audio traffic (but you've already described how there was no audio, this would merely verify that the audio was not arriving at the server).
There *may* be traffic in the sip debug entries in the asterisk cli that provides a hint of what's happening. But if the audio from the carrier is "just stopping", there would not be an entry unless they are doing this on purpose and provided a message such as "payment required". Unlikely if other calls are not affected.
You could attempt to jump into the agent's session by monitoring it, and verifying that you can the agent can hear each other. This would provide proof that the agent's connection to the server is not broken and that the meetme room is not broken.
But if you can't get a new carrier just by signing up ... you must not be in the US.
williamconley wrote:SIP Connections to carriers are not running engines. Each time a call is initiated, a new connection is made. Each time a call terminates, the connection is broken.
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