Outbound Calling - long delay to connect

[rbudnikas]

Hello Everyone,

I'm new to 3CX and reviewed similar questions on this forum to find the solution to my issue. My configuration is 3CX/CallCentric with an Aastra 9133i handset and android mobile phone. Both behave the same way. I can make extension calls without any problems and within seconds the corresponding handset rings. I can even get a quick ring outside of the local network. The issue is that when I call any number from the Aastra or the android phone it takes a significant amount of time. Please see the log below.

If you notice at 13-Nov-2012 16:17:15.814 it states, "Currently active calls - 1: [91]" but it is still ringing until 13-Nov-2012 16:17:47.817. That is 32 seconds before the actual phone rings which is actually 45 seconds total time from when the call was actually initiated by the handset. That is a significant amount of time to pass. Does anyone have any ideas?

Thanks,

Robert

13-Nov-2012 16:17:54.506 Leg L:91.2[Line:10000>>19054078041] is terminated: Cause: 487 Request Terminated/INVITE from 204.11.192.169:5060
13-Nov-2012 16:17:54.347 [CM503008]: Call(C:91): Call is terminated
13-Nov-2012 16:17:54.344 Leg L:91.1[Extn] is terminated: Cause: CANCEL from 192.168.1.2:1026
13-Nov-2012 16:17:47.817 Currently active calls - 1: [91]
13-Nov-2012 16:17:15.814 Currently active calls - 1: [91]
13-Nov-2012 16:17:03.408 [CM505003]: Provider:[CallCentric - US] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:[email protected]:49152]
13-Nov-2012 16:17:02.241 [CM503025]: Call(C:91): Calling T:Line:10000>>19054078041@[Dev:sip:[email protected]:5060] for L:91.1[Extn]
13-Nov-2012 16:17:02.157 [CM503027]: Call(C:91): From: Extn:101 ("101" <sip:[email protected]:5060>) to T:Line:10000>>9054078041@[Dev:sip:[email protected]:5060]
13-Nov-2012 16:17:02.157 [CM503004]: Call(C:91): Route 3: from L:91.1[Extn] to T:Line:10000>>9054078041@[Dev:sip:[email protected]:5060]
13-Nov-2012 16:17:02.157 Line limit check: Current # of calls for line Lc:10000(@CallCentric - US[<sip:[email protected]:5060>]) is 0; limit is 2
13-Nov-2012 16:17:02.157 [CM503027]: Call(C:91): From: Extn:101 ("101" <sip:[email protected]:5060>) to T:Line:10000>>9054078041@[Dev:sip:[email protected]:5060]
13-Nov-2012 16:17:02.157 [CM503004]: Call(C:91): Route 2: from L:91.1[Extn] to T:Line:10000>>9054078041@[Dev:sip:[email protected]:5060]
13-Nov-2012 16:17:02.157 Line limit check: Current # of calls for line Lc:10000(@CallCentric - US[<sip:[email protected]:5060>]) is 0; limit is 2
13-Nov-2012 16:17:02.157 [CM503027]: Call(C:91): From: Extn:101 ("101" <sip:[email protected]:5060>) to T:Line:10000>>19054078041@[Dev:sip:[email protected]:5060]
13-Nov-2012 16:17:02.157 [CM503004]: Call(C:91): Route 1: from L:91.1[Extn] to T:Line:10000>>19054078041@[Dev:sip:[email protected]:5060]
13-Nov-2012 16:17:02.157 Line limit check: Current # of calls for line Lc:10000(@CallCentric - US[<sip:[email protected]:5060>]) is 0; limit is 2
13-Nov-2012 16:17:02.157 Call(C:91): Call from Extn:101 to 9054078041 matches outbound rule 'Rule for CallCentric - US'
13-Nov-2012 16:17:02.155 [CM503001]: Call(C:91): Incoming call from Extn:101 to "9054078041" <sip:[email protected]:5060>

 

[farsight]

It looks like you have already ID'd the issue. It's the SIP trunk. Either the provider or the setup in 3CX.

 

[rbudnikas]

Ok, so I've id'd the issue. My question where is the problem, on the 3cx side or Callcentric? Initially I thought it was due to the 3 routes in the outbound rule, but if I reduce the routes to a single one, I get a message saying that the number I am dialing is not assigned. I don't get the message with a minimun of 2 routes. Í've checked for latency between the server and call centric and it is at average, 33ms so I can say that it might be the problem with the transfer between 3cx to call centric.

Any suggestions would be most helpful.

Thanks,

Robert

 

[leejor]

Given the recent issues, brought up in the forum, it may very well be your provider, or your settings and how they cause 3CX to interact with them. You might try a very simple, single trunk set-up, to see if the delay is still there after the call is passed to Callcentric. You may also want to consider making use of Wireshark for more detailed information about the interaction between 3CX and Callcentric. The results would probably provide a definitive answer as to where the delay is occurring. If it shows that the wait occurs after the call is passed to Callcentric, you can then use that information to contact them for help in resolving the problem.

 

[jpillow]

I'd be willing to bet $100 to $1 its your provider as you're using Callcentric they've had a lot of issues lately includeing their servers being down. Also their support is non existent I'd move to another provider, either do that or wait it our.

 

[markshehan]

I am going to say it is your call provider.

You can draw your own conclusions about them based on your experience and what others in here are saying.

If you run a wireshark on an outbound call you will see your sip timers firing (meaning they sent the invite to the provider but had no response, so there are timers that increase in duratation (starting at 0.5 seconds) until a 30 second time out. So you will see lots of invites out, but no replies to them.

So other than a configuration problem on your end, it means their servers are either offline or cannot cope with the load.

If you switch provider, make sure they have high availability servers, balanced loads and do not confine you to a single server to register/place calls. Ideally they should have multiple servers in each data center, load balanced and using SRV records for you to connect too and also geographically dispersed data centers that auto failover should one go totally offline. And you should not have to do anything to your 3cx should any of their systems fail or get overloaded. Your performance should be identical.

 

[markshehan]

I am going to say it is your call provider.

You can draw your own conclusions about them based on your experience and what others in here are saying.

If you run a wireshark on an outbound call you will see your sip timers firing (meaning they sent the invite to the provider but had no response, so there are timers that increase in duratation (starting at 0.5 seconds) until a 30 second time out. So you will see lots of invites out, but no replies to them.

So other than a configuration problem on your end, it means their servers are either offline or cannot cope with the load.

If you switch provider, make sure they have high availability servers, balanced loads and do not confine you to a single server to register/place calls. Ideally they should have multiple servers in each data center, load balanced and using SRV records for you to connect too and also geographically dispersed data centers that auto failover should one go totally offline. And you should not have to do anything to your 3cx should any of their systems fail or get overloaded. Your performance should be identical.

 

[sigma1]

Over the last month we had almost 200 new clients coming from Callcentric for similar issues (and poor voice quality) and this is inline with what we have seen. They have been having serious infrastructure trouble, I am sorry "fake hacking". In any event, don't waste your time and switch to any other SIP trunk provider and you will be all set. Keep in mind that due to their non-existing support, it takes a while to get an FOC to port your numbers from them.