Call being dropped after 15 minutes [RESOLVED]

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Call being dropped after 15 minutes [RESOLVED]

Postby meneses » Mon Oct 15, 2012 8:10 am

Hello,

I'm having a weird problem, tried a few possible solutions with no success and now I need help from you guys.

Scenario:
A have three asterisk boxes one as E1 gateway and the other two as a PBX machines, all of them connected through SIP trunks. So far so good.

Problem:
SIP calls between the PBX machines are fine, but all inbound calls (outbound calls are fine) coming from the asterisk E1 gateway are being dropped by the PBX machines after 15 minutes and the following warning can be seen on the logs:

Code: Select all
p-retransmit.txt.
[Oct  4 11:05:43] WARNING[6066] chan_sip.c: Maximum retries exceeded on transmission 636c243a08a8d9bd1154c1ed0dfd53f3@10.145.80.10 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt.
[Oct  4 11:05:43] WARNING[6066] chan_sip.c: Hanging up call 636c243a08a8d9bd1154c1ed0dfd53f3@10.145.80.10 - no reply to our critical packet (see doc/sip-retransmit.txt).
[Oct  4 11:05:43] VERBOSE[16590] logger.c:     -- Executing [h@macro-dial:1] Macro("SIP/TRONCO-85-00000d61", "hangupcall") in new stack
[Oct  4 11:05:43] VERBOSE[16590] logger.c:     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/TRONCO-85-00000d61", "1?skiprg") in new stack


What has been tried:

- I fiddled a little with the t1min parameter (from 200 to 800) but the problem persists.
- Disabled the qualify param.

Trunk config the same on both sides, execpt the IP adress and trunk name:

Code: Select all
[TRONCO-85]
type=friend
qualify=no
host=10.145.80.10
disallow=all
allow=alaw&ulaw&h264
context=from-internal


General section from sip.conf:
Code: Select all
[general]
videosupport=yes
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
qualifyfreq=120
callcounter=yes
limitonpeers=yes
rtptimeout=240
rtpholdtimeout=600



Data (for all machines):
asterisk 1.6.0.26
dahdi 2.3.0.1
libpri 1.4.10.2

It's old, I know, but update is not an option for now :/

Wrapping up
What may be causing that kind of behaviour? Calls being dropped after 15 minutes ONLY if they came from the E1 gateway asterisk and only the inbound calls.
Last edited by meneses on Wed Jun 03, 2015 4:45 am, edited 1 time in total.
meneses
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Posts: 72
Joined: Fri Jul 25, 2008 7:26 am

Re: Call being dropped after 15 minutes

Postby malcolmd » Mon Oct 15, 2012 8:56 am

SIP Session timers turned on?
Malcolm Davenport
Digium, Inc. | Senior Product Manager
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Re: Call being dropped after 15 minutes

Postby meneses » Mon Oct 15, 2012 10:12 am

It is not set, therefore it defaults to 'accept requests', but never originate.

Here's the result of 'sip show settings'

Code: Select all
Global Settings:
----------------
  UDP SIP Port:           5060
  UDP Bindaddress:        0.0.0.0
  TCP SIP Port:           Disabled
  TLS SIP Port:           5061
  TLS Bindaddress:        127.0.0.1
  Videosupport:           Yes
  Textsupport:            No
  AutoCreate Peer:        No
  Ignore SDP sess. ver.:  No
  Match Auth Username:    No
  Allow unknown access:   Yes
  Allow subscriptions:    Yes
  Allow overlap dialing:  Yes
  Allow promsic. redir:   No
  Enable call counters:   Yes
  SIP domain support:     No
  Realm. auth:            No
  Our auth realm          asterisk
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    Yes
  Call limit peers only:  Yes
  Direct RTP setup:       No
  User Agent:             Asterisk PBX 1.6.0.26-FONCORE-r78
  SDP Session Name:       Asterisk PBX 1.6.0.26-FONCORE-r78
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Caller ID:              Unknown
  From: Domain:           
  Record SIP history:     Off
  Call Events:            Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          -1
  SIP realtime:           Disabled
  Qualify Freq :          120000 ms

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS3
  IP ToS RTP audio:       EF
  IP ToS RTP video:       AF41
  IP ToS RTP text:        CS0
  802.1p CoS SIP:         4
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   6
  802.1p CoS RTP text:    5
  Jitterbuffer enabled:   No
  Jitterbuffer forced:    No
  Jitterbuffer max size:  -1
  Jitterbuffer resync:    -1
  Jitterbuffer impl:     
  Jitterbuffer log:       No

Network Settings:
---------------------------
  SIP address remapping:  Disabled, no localnet list
  Externhost:             <none>
  Externip:               0.0.0.0:0
  Externrefresh:          10
  Internal IP:            127.0.0.1:5060
  STUN server:            0.0.0.0:0

Global Signalling Settings:
---------------------------
  Codecs:                 0x10000e (gsm|ulaw|alaw|h263p)
  Codec Order:            alaw:20,ulaw:20,gsm:20
  Relax DTMF:             No
  RFC2833 Compensation:   No
  Compact SIP headers:    No
  RTP Keepalive:          0 (Disabled)
  RTP Timeout:            240
  RTP Hold Timeout:       600
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         Yes
  Pedantic SIP support:   No
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
  Notify hold state:      Yes
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No
  Outb. proxy:            <not set>
  Session Timers:         Accept
  Session Refresher:      uas
  Session Expires:        1800 secs
  Session Min-SE:         90 secs
  Timer T1:               500
  Timer T1 minimum:       100
  Timer B:                32000
  No premature media:     No

Default Settings:
-----------------
  Context:                from-sip-external
  Nat:                    RFC3581
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               pt_BR
  MOH Interpret:          default
  MOH Suggest:           
  Voice Mail Extension:   *97

----
meneses
Oldsterisk
 
Posts: 72
Joined: Fri Jul 25, 2008 7:26 am

Re: Call being dropped after 15 minutes

Postby malcolmd » Mon Oct 15, 2012 11:00 am

So if you do a sip show channel <xyz> on the call, do you see a timer active that would expire at 15 minutes?
Malcolm Davenport
Digium, Inc. | Senior Product Manager
malcolmd
Moves Like Spencer
 
Posts: 3019
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Location: Huntsville, AL, US

Re: Call being dropped after 15 minutes

Postby meneses » Mon Oct 15, 2012 12:34 pm

No, i dont. But just in case I'm missing something, here's the output:

E1 Gateway <---> PBX

Code: Select all
  * SIP Call
  Curr. trans. direction:  Outgoing
  Call-ID:                3b04171063139c67468865cd3b235399@10.145.80.10
  Owner channel ID:       SIP/TRONCO-84-00001168
  Our Codec Capability:   12
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   12
  Joint Codec Capability:   12
  Format:                 0x8 (alaw)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    10.144.80.10:5060
  Received Address:       10.144.80.10:5060
  SIP Transfer mode:      open
  NAT Support:            RFC3581
  Audio IP:               10.145.80.10 (local)
  Our Tag:                as433f0ecd
  Their Tag:              as53493711
  SIP User agent:         Asterisk PBX 1.6.0.26-FONCORE-r78
  Username:               8412
  Peername:               TRONCO-84
  Original uri:           sip:8412@10.144.80.10
  Need Destroy:           No
  Last Message:           Tx: ACK
  Promiscuous Redir:      No
  Route:                  sip:8412@10.144.80.10
  DTMF Mode:              rfc2833
  SIP Options:            (none)
  Session-Timer:          Active
  S-Timer Interval:       1800
  S-Timer Refresher:      uas
  S-Timer Expirys:        0
  S-Timer Sched Id:       1636213
  S-Timer Peer Sts:       Active
  S-Timer Cached Min-SE:  0
  S-Timer Cached SE:      0
  S-Timer Cached Ref:     auto
  S-Timer Cached Mode:    Accept



PBX <---> E1 gateway

Code: Select all
  * SIP Call
  Curr. trans. direction:  Incoming
  Call-ID:                3b04171063139c67468865cd3b235399@10.145.80.10
  Owner channel ID:       SIP/TRONCO-85-0000091d
  Our Codec Capability:   12
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   12
  Joint Codec Capability:   12
  Format:                 0x8 (alaw)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    10.145.80.10:5060
  Received Address:       10.145.80.10:5060
  SIP Transfer mode:      open
  NAT Support:            RFC3581
  Audio IP:               10.144.80.10 (local)
  Our Tag:                as53493711
  Their Tag:              as433f0ecd
  SIP User agent:         Asterisk PBX 1.6.0.26-FONCORE-r78
  Peername:               TRONCO-85
  Original uri:           sip:  6870@10.145.80.10
  Caller-ID:                6870
  Need Destroy:           No
  Last Message:           Rx: ACK
  Promiscuous Redir:      No
  Route:                  sip:  6870@10.145.80.10
  DTMF Mode:              rfc2833
  SIP Options:            replaces replace timer
  Session-Timer:          Active
  S-Timer Interval:       1800
  S-Timer Refresher:      uas
  S-Timer Expirys:        0
  S-Timer Sched Id:       2274159
  S-Timer Peer Sts:       Active
  S-Timer Cached Min-SE:  90
  S-Timer Cached SE:      0
  S-Timer Cached Ref:     uas
  S-Timer Cached Mode:    Accept


PBX <--> Telephone

Code: Select all
  * SIP Call
  Curr. trans. direction:  Outgoing
  Call-ID:                530c6cba1f794d5d1237bdac28c97eeb@10.144.80.10
  Owner channel ID:       SIP/8412-0000091e
  Our Codec Capability:   12
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   8
  Joint Codec Capability:   8
  Format:                 0x8 (alaw)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    10.140.84.22:5062
  Received Address:       10.140.84.22:5062
  SIP Transfer mode:      open
  NAT Support:            Always
  Audio IP:               10.144.80.10 (local)
  Our Tag:                as3527cea2
  Their Tag:              1723306290
  SIP User agent:         Yealink SIP-T22P 7.61.0.80
  Username:               8412
  Peername:               8412
  Original uri:           sip:8412@10.140.84.22:5062
  Need Destroy:           No
  Last Message:           Tx: ACK
  Promiscuous Redir:      No
  Route:                  sip:8412@10.140.84.22:5062
  DTMF Mode:              rfc2833
  SIP Options:            (none)
  Session-Timer:          Inactive


Thanks for the help, btw
meneses
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Posts: 72
Joined: Fri Jul 25, 2008 7:26 am

Re: Call being dropped after 15 minutes

Postby malcolmd » Mon Oct 15, 2012 12:48 pm

On both servers, go ahead and set session-timers=refuse in the [general] section of sip.conf and let's see what happens.
Malcolm Davenport
Digium, Inc. | Senior Product Manager
malcolmd
Moves Like Spencer
 
Posts: 3019
Joined: Wed Aug 03, 2005 3:53 pm
Location: Huntsville, AL, US

Re: Call being dropped after 15 minutes

Postby meneses » Wed Oct 17, 2012 7:25 am

It worked, malcolmd.

After seting 'session-timers=refuse' in the [general] section of sip.conf, just like you said, there where no hangups after 15min.

Thanks for the help.

But, if no one where originating sip-timers, how come this could be a problem?
meneses
Oldsterisk
 
Posts: 72
Joined: Fri Jul 25, 2008 7:26 am

Re: Call being dropped after 15 minutes

Postby malcolmd » Wed Oct 17, 2012 7:39 am

Session timer support in Asterisk was fairly extensively re-worked, to work properly, in the most recent versions, so all bets are off for previous versions. :D
Malcolm Davenport
Digium, Inc. | Senior Product Manager
malcolmd
Moves Like Spencer
 
Posts: 3019
Joined: Wed Aug 03, 2005 3:53 pm
Location: Huntsville, AL, US


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