vicidial.org

Good day. I would like to ask for your assistance regarding my drop calls problem.

The scenario is that, an agent is calling using manual dial. After some connected calls and conversations with the client, the agent suddenly cannot hear any response from the client. The client surely didn't hang up because there is no hangup sound heard by the agent. On vicidial realtime monitoring report, the said agent status is incall. I don't know what you call this situation but I used the term drop calls. Just correct me if I'm wrong.

Our server cpu load is not that high and our bandwidth / internet consumption is around 2mbps only out of 4mbps fiber optic.

Vicidial version is VERSION: 2.6-395a BUILD: 130221-1736 and our Asterisk version is 1.4.44-vici build 13

probably with your voip? is this happening before? do a traceroute using mtr to your voip server ip.

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.

rrb555 wrote:probably with your voip? is this happening before? do a traceroute using mtr to your voip server ip.

This is not happening before. I did traceroute to my carriers ip address and it seems fine. below 1% packet loss.

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.

3. This is not fresh install. Its been installed around 3 years already if I'm not mistaken. And no new carriers.

4. I don't have a second carrier.

6. Some agents experiencing drop calls. The thing is, their calls are good for around 5-10 mins and then it's gone.

I already tried reloading and restarting asterisk, also restarted servers.

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: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.

Ahhh... Wait, is there a way to trace / check the dropcalls under asterisk logs? or in the debug? Sorry for such a newbie.

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: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

Ok. The thing is, even though I apply for a second carrier, it would take time in order to be up and running. So i'm trying to figure it out without second carrier if ever.

So there is another option which is the audio is just silenced and the call didn't drop?

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: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.

Is there any other way to restart my connection to my carrier?

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.

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.

From our conversations, basically my best shot in solving this problem is adding a new carrier.

I'll try to contact my carrier and see if they can solve the problem or blame it on our network

You're a Genius. I wish I'd thought of that!

(Had to do it, LOL).