[SR-Users] Calls remain stuck in state 1

[Daniel-Constantin Mierla]

Hello,

3.1 is rather old for dialog module, you should upgrade to more recent 
version because dialog got lot of work.

The general hint is that you have to create the dialog as last operation 
before t_relay().

Cheers,
Daniel

On 5/24/13 11:05 AM, Giany wrote:
> Hello,
>
> We are using dipatcher to limit the concurrent number of calls, 
> problem is that from time to time
>  calls remain stuck in state 1 and it breaks our concurrent limits..I 
> was not able to make a kamailio
> log with high debug as it happens randomly. Attached is a tcpdump flow:
>
> Conv.| Time    | serverA                         | Provider           
>          |
>      |       |                   | RemoteEnd     |
> 112  |938.355  |         INVITE SDP (g729 g711U GSM X-NSERTPType-100 
> te...hone-eventRT)
>      |       |(5060)   ------------------>  (5050)   |             |
> 112  |938.357  |         100 Trying|                   |               |
>      |       |(5050)   ------------------>  (5060)   |             |
> 113  |938.420  |         INVITE SDP (g711U g729 
> telephone-eventRTPType-...)|
>      |       |(5050)   ------------------>  (5060)   |             |
> 113  |938.422  |         100 trying -- your call is important to us   
>      |
>      |       |(5060)   ------------------>  (5050)   |             |
> 113  |938.422  |         INVITE SDP (g711U g729 
> telephone-eventRTPType-...)|
>      |       |(5060)   ------------------>  (1416)   |             |
> 113  |938.908  |         INVITE SDP (g711U g729 
> telephone-eventRTPType-...)|
>      |       |(5060)   ------------------>  (1416)   |             |
> 113  |939.908  |         INVITE SDP (g711U g729 
> telephone-eventRTPType-...)|
>      |       |(5060)   ------------------>  (1416)   |             |
> 113  |941.906  |         INVITE SDP (g711U g729 
> telephone-eventRTPType-...)|
>      |       |(5060)   ------------------>  (1416)   |             |
> 113  |945.907  |         INVITE SDP (g711U g729 
> telephone-eventRTPType-...)|
>      |       |(5060)   ------------------>  (1416)   |             |
> 112  |948.358  |         CANCEL    |                   |               
> |SIP Request
>      |       |(5060) <--------------------------------------  (61016)  |
> 112  |948.359  |         CANCEL    |                   |               
> |SIP Request
>      |       |(5060)   ------------------>  (5050)   |             |
> 112  |948.359  |         200 canceling                 |               
> |SIP Status
>      |       |(5060) -------------------------------------->  (61016)  |
> 112  |948.359  |         487 Request Terminated          |             
>   |SIP Status
>      |       |(5050)   ------------------>  (5060)   |             |
> 112  |948.359  |         200 OK    |                   |               
> |SIP Status
>      |       |(5050)   ------------------>  (5060)   |             |
> 112  |948.359  |         ACK       |                   |               
> |SIP Request
>      |       |(5060)   ------------------>  (5050)   |             |
> 112  |948.360  |         487 Request Terminated        |               
> |SIP Status
>      |       |(5060) -------------------------------------->  (61016)  |
> 113  |948.360  |         CANCEL    |                   |               
> |SIP Request
>      |       |(5050)   ------------------>  (5060)   |             |
> 113  |948.360  |         200 canceling                 |               
> |SIP Status
>      |       |(5060)   ------------------>  (5050)   |             |
> 112  |948.365  |         ACK       |                   |               
> |SIP Request
>      |       |(5060) <--------------------------------------  (61016)  |
>
>
> dialog::  hash=136:689416016
>          state:: 1
>          ref_count:: 3
>          timestart:: 0
>          timeout:: 0
>          callid:: 745eed805cad6b9a3e1727d169cf3461 at serverA:5050
>          from_uri:: sip:fromnumber at serverA:5050
>          from_tag:: as53f6bee4
>          caller_contact:: sip:fromnumber at serverA:5050
>          caller_cseq:: 102
>          caller_route_set::
>          caller_bind_addr:: udp:serverA:5060
>          callee_bind_addr::
>          to_uri:: sip:internalnr at serverA:5060
>          to_tag::
>          callee_contact::
>          callee_cseq::
>          callee_route_set::
> As you see the remoteEnd does not answer at all to this request (due to network issue most likely)
> and the provider sends a CANCEL after approx 3 seconds.
>  From what I see the INVITE that is sent from asterisk towards kamailio
>   remains stuck(938.420).
> We are using Kamailio 3.1.6.Any idea what could be the reason for this?
>
> Thank you.
>
>
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-- 
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
   * http://asipto.com/u/katu *

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