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:
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:
[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:
[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.
SIP Session timers turned on?
It is not set, therefore it defaults to ‘accept requests’, but never originate.
Here’s the result of ‘sip show settings’
[code]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:
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:
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
[/code]
So if you do a sip show channel on the call, do you see a timer active that would expire at 15 minutes?
No, i dont. But just in case I’m missing something, here’s the output:
E1 Gateway <—> PBX
* 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
* 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
* 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
On both servers, go ahead and set session-timers=refuse in the [general] section of sip.conf and let’s see what happens.
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?
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. 