Calls Drop Randomly

[Jimc]

We are having an issue with our Switchvox system (5.5.2) and randomly dropped calls. We are using Optimum IP-SIP for our service.

On a random basis, calls will suddenly disconnect while in progress. Sometimes the logs show dropped by the SIP provider, other times by us and yet other times by the caller. We have had these traced by the provider and they claim nothing is wrong on their side. However, I do not believe this to be the case, as we show in the advanced error logs that there are quite a few times that their equipment has failed to authenticate our system. I have no proof that this is causing the dropped calls, but am suspect. Some callers have told us that right before the call drops, they lose audio with us.

My Questions are:

1. has anyone else experienced a similar problem? And How did you resolve it?

2. Is it more reliable to go back to SIP over a PRI Card?

[intellectit]

Have you tried 5.5.3 yet?

Switchvox @ Intellect IT

[Jimc]

Not yet, downloaded it today and will apply tonight.

[barrangatan]

Also - are internal extension to extension calls being dropped also when it happens? If they are, then the issue may not just be provider-related. And if the issue is happening at least once a day, I'd suggest running the Asterisk CLI debug under the 'Advanced Debugging' section of the Admin Portal for a day, then note the time(s) that the calls have dropped, and contact Switchvox support so they can look into the packet captures.

[brucexx]

We were having problems with Optimum SIP for 4 months with one of our clients. Every time they said it was on our end. It turned out that they have very limited way of troubleshooting this type of issues. I had to literary tell them that they Edgemark device (media gateway) is not responding to the requests on the local network as they were telling me everything is ok. The problem was with over-utilization of a "node" in the area. They had to "split the node" whatever that means and since then it is solid.

The call log is not the best place to look for answers with randomly dropped call. Consider a scenario in which internal caller stops hearing the other person (RTP packets are not being sent anymore from Edgemark), the internal caller would think that the call dropped and will disconnect, the call log will show you that the call was disconnected properly by the system.

Setup a network trace. If high call volume do port mirroring on the switch and dedicated workstation to capture problematic calls. I am afraid that this is the only way to be sure.

We are running Optimum SIP on 4.6.4, 5.1.2, 5.5, 5.5.2 and 5.5.3 - and it is solid on switchvox side, every problem we had with Optimum was related to the service and not the system (assuming the SIP trunks are configured properly).

[Jimc]

I had one of their engineers basically tell me that the other day. We switched back to PRI and are having the exact same problem. Time to go back to them again..

[brucexx]

If you switch back to PRI it will not solve your problem if this is the overloaded node. The calls are going into the same place anyway being terminated on Edgemark. I would never go back to PRI because of the ability of troubleshooting the PRI voice and signaling is limited on Switchvox. There is PRI log and it is not bad but I prefer network trace with RTP and SIP. I would go back to SIP and start tracing - this is really the only way to show them exactly what is going on.

[johnwells]

If you are willing to try a different SIP provider, try Nexvortex. We've been using them for about a year and a half, and they have been amazing. To be honest, we have had three outages with them.

The first was about 10 minutes, and they warned us a few minutes ahead of time (one of their carriers was having an issue).
The second was nearly 4 hours. During the big ice storm, their local LEC lost power after their generators ran out of fuel, but I have been assured that situation has been addressed.
The third time was a little less than 15 minutes, and I don't remember what caused that one.

Considering all the issues we had with Bandwidth.com before switching, I consider these folks phenomenal.

If you call, ask for Mike Nowak, and let him know John Wells sent you.

[brucexx]

Is this service provided over the public internet ?

Optimum actually puts device on your local network and from there they are providing QoS so no quality problem at all - when it works

[Jimc]

The only problem with going with any other carrier, is that we are stuck with Optimum for our data at this location, as there is no other provider available. And since I am sure that this is a capacity problem, I doubt this will help the situation. I have given our sales rep a earful today, and hoping this moves it along. I know that this is a node issue, as I have had techs tell us this a few times.

[brucexx]

Optimum when works is solid and provides QoS for SIP over their Vocie network - which can be a problem if you are using SIP provider that uses public internet.

Splitting the node takes time though - it took them like 2 months to do this.

[Jimc]

I just had their people here today, they think they are going to have to split the node, they put traps on the lines to monitor the activity, but since most of the drops happen at peak times, they are leaning to it being a over subscribed node.

[ve2tmq]

I have the same issue with asterisk 1.8.18.0 compiled on Linux Slackware. In communication suddenly call drops. No noice, no click, just silence.

[adamgoodfriend]

We have had nothing but problems with optimum voice sip. We had a customer that was working fine then after 2 month their sip became unusable with choppy distorted voice. We called cablevision and reported the problem and they kept blaming the pbx. We ran multiple test (including adding and internet sip trunk that worked flawlessly) and determined it was the Edgewater device that they installed. After 2 weeks of fighting with cablevision (and a customer with no usable phone) they finally agreed to replace the device. The service worked after replacing the device. Well some time after that we the issue popped up again and we cancled the service and went with broadvox. Do not be fooled into thinking that because cablevision is your isp that somehow your voice quality will be better. The truth is they have no idea what they are doing when it comes to sip and their edgewater devices are garbage. If your are in a cablevison area the chances are you have a broadvox point of presence near you and you will not have any quality issues with your sip even with out cablevisions "QOS". I have many customers that have over 10 trunks (concurrent sessions) and run them over cablevision ISP with no jitter or voice problems. If you absolutely must have "QOS" all the way to the PSTN you should consider getting an MPLS connection to a real trunking provider. Moral of the story is STAY AWAY FROM OPTIMUM VOICE SIP! Also our customer made sure the sales rep who sold them the sip services was fired.
-Adam Goodfriend
Elevated Computing

[hiltonmw]

Not that this helps you, Adam, but there's been a Known Defect between Digium Switchvox and Edgewater devices since October:

http://www.digium.com/en/products/switc ... hvox-today