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Posted

Every once in a while a few users complain that they can't answer incoming calls. From their point of view they pick up the handset and it's like there was no call. I was able to trace 3 failed calls in two different domains (and kept detailed pcap traces and logs for the last one of them).

SIP trace with carrier
======================
|Time     | carrier GW                            |
|         |                   |          PBX      |                   
|0,000    |         INVITE SDP                    |SIP From: Anonymous ...
|         |(5060)   ------------------>  (5060)   |
|0,002    |         100 Trying|                   |SIP Status
|         |(5060)   <------------------  (5060)   |
|0,010    |         200 Ok SDP                    |SIP Status
|         |(5060)   <------------------  (5060)   |
|0,032    |         ACK       |                   |SIP Request
|         |(5060)   ------------------>  (5060)   |
|11,980   |         BYE       |                   |SIP Request
|         |(5060)   <------------------  (5060)   |
|11,990   |         200 OK    |                   |SIP Status
|         |(5060)   ------------------>  (5060)   |

SIP trace with VoIP phone
=========================
|Time     | PBX                                   |
|         |                   |     VoIP Phone    |                   
|0,000    |         INVITE SDP                    |SIP From: "Anonymous"...
|         |(5060)   ------------------>  (1036)   |
|0,053    |         100 Trying|                   |SIP Status
|         |(5060)   <------------------  (1036)   |
|0,185    |         180 Ringing                   |SIP Status
|         |(5060)   <------------------  (1036)   |
|5,482    |         200 OK SDP                    |SIP Status
|         |(5060)   <------------------  (1036)   |
|5,483    |         ACK       |                   |SIP Request
|         |(5060)   ------------------>  (1036)   |
|5,547    |         BYE       |                   |SIP Request
|         |(5060)   ------------------>  (1036)   |
|5,626    |         200 OK    |                   |SIP Status
|         |(5060)   <------------------  (1036)   |

So the PBX got a 200 OK from the phone and a few milisecs later it send a BYE back to the phone and to the carrier dropping the call. Looking at the logs to see why I see this:

[8] 20140716134853: Call state for call object 143211: alerting
[...]
[...]
[8] 20140716134854: call port 22: state code from 100 to 200
[8] 20140716134854: call port 20: state code from 200 to 200
[7] 20140716134854: 14675947-3614496522-991048@carrier: RTP pass-through mode
[7] 20140716134854: 1e4c6746@pbx: RTP pass-through mode
[7] 20140716134854: Determine pass-through mode after receiving response
[8] 20140716134854: Call state for call object 143211: connected
[...]
[8] 20140716134854: call port 20: state code from 200 to 486
[8] 20140716134854: call port 22: state code from 200 to 486
[...]
[...]
[7] 20140716134855: Call 14675947-3614496522-991048@carrier: Clear last request
[5] 20140716134855: BYE Response: Terminate 14675947-3614496522-991048@carrier
[7] 20140716134855: 1e4c6746@pbx: Media-aware pass-through mode
[8] 20140716134855: Clearing call port 20, SIP call id 14675947-3614496522-991048@carrier
[8] 20140716134855: Remove leg 347938: call port 20, SIP call id 14675947-3614496522-991048@carrier
[7] 20140716134855: Call 1e4c6746@pbx: Clear last request
[5] 20140716134855: BYE Response: Terminate 1e4c6746@pbx
[8] 20140716134855: Clearing call port 22, SIP call id 1e4c6746@pbx
[8] 20140716134855: Remove leg 347940: call port 22, SIP call id 1e4c6746@pbx

I've noticed that call port 22 that correspond to this call goes from 100 to 200 and almost immediately after that from 200 to 486. I guess these numbers correspond to SIP status so the 486 (busy) seems really strange to me. This (BYE send from pbx and log lines like the above) happened in all 3 cases that I was able to follow closely.

 

I can provide the pcap traces and full log if needed (I just can't post them publicly)

Posted

Yea it should also complain about that in the logs. However you will see it very clear in the SIP messages, potentially a PCAP or just the regular log with the SIP packets inside. If you want to share logs, open a ticket on zendesk (the Vodia support system).

  • 1 year later...
Posted

Is there any resolution on this topic, as we just starting having exactly the same experience. The logs are very similar, but it doesnt happen after the ack, it happens well into the call.

 

I noticed it would happen on a second call, after the recipient had disconnected the first call.

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