Hi all,
My environment is :
[Customer’s Predictif application (VOXCO)] <–PRI Line–> [Asterisk Box-1] <–VoIP SIP–> [Asterisk Box-2]
<–VoIP (IAX)–> [Asterisk Box-3] ← VoIP(SIP)–> [Telco Operator]
- Asterisk Box-1/2/3 : v 1.6.2.9 + Dahdi 2.3
- Digium Card TE420 (5th Gen) on Asterisk Box-1
- Transcoding ulaw to g729 on Asterisk Box-1
- Number Calls simultaneous generate by VOXCO : no limit and can be 5, 15, 20 or 25, …
- Latency between Asterisk Box-1 and Asterisk Box-2 : 7ms
- Latency between Asterisk Box-2 and Asterisk Box-3 : 200ms
- Latency Asterisk Box-3 abd Telco Operator is about 8ms
- Bandwidth dedicated between Box-1 to Telco Operator and no congestion
- No simultaneous call limit with our Telco operator
My problem :
- Voxco launches new calls simultaneous and some new calls FAILED, even if with only ~8 calls active simultaneous
But when I call, a phone number who is FAILED, directly from Voxco or from sip-phone configured on the Asterisk Box-1, I never had this problem.
1- My brief config on the Asterisk Box-1 :
- timing : res_timing_dahdi.so
- extension.conf : exten => _00X.,1,Dial(SIP/to-asterisk-box-2/${EXTEN:2},240)
- sip show settings :
Global Settings:
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: Disabled
TLS SIP Port: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promsic. redir: No
Enable call counters: No
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: No
Direct RTP setup: No
User Agent: Asterisk PBX 1.6.2.9-2+squeeze3
SDP Session Name: Asterisk PBX 1.6.2.9-2+squeeze3
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Network QoS Settings:
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
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: Yes
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: x.x.x.x:5060
STUN server: 0.0.0.0:0
Global Signalling Settings:
Codecs: 0x10c (ulaw|alaw|g729)
Codec Order: g729:20,ulaw:20,alaw:20
Relax DTMF: No
RFC2833 Compensation: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
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
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Refuse
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: Yes
Default Settings:
Allowed transports: UDP
Outbound transport: UDP
Context: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: fr_FR
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
-
dahdi/system.conf :
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31 -
asterisk/dahdi-channels.conf :
switchtype=euroisdn
context=predictif
group=1
echocancel=yes
signalling=pri_cpe
channel =>1-15,17-31 -
My config iax2 on Asterisk Box-2 :
[general]
bindport=4569
bindaddr=x.x.x.x
iaxcompat=no
language=en
bandwidth=low
jitterbuffer=no
forcejitterbuffer=no
iaxthreadcount = 10
iaxmaxthreadcount = 100
autokill=yes
[box2-tobox3]
type = peer
username = predictif-users
secret = YTRESDF
host = y.y.y.y
qualify = yes
trunk = yes
requirecalltoken = no
disallow = all
allow = g729
jitterbuffer = yes
transfer=no
2- Duration call by asterisk logs :
- Asterisk Box-1 : 12s (status FAILED)
- Asterisk Box-2 : 1s (status FAILED)
- Asterisk Box-3 : 0s (status FAILED)
3- Debug on Asterisk Box-1
- pri debug :
Message Type: RELEASE COMPLETE (90)
Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1)
Ext: 1 Cause: Circuit/channel congestion (34), class = Network Congestion (resource unavailable) (2) ]
- sip debug :
DEBUG[21024] chan_sip.c: Header 0 [ 31]: SIP/2.0 503 Service Unavailable
DEBUG[21024] chan_sip.c: Header 12 [ 50]: X-Asterisk-HangupCause: Circuit/channel congestion
DEBUG[21024] chan_sip.c: Header 13 [ 30]: X-Asterisk-HangupCauseCode: 34
DEBUG[18852] channel.c: Hanging up channel ‘SIP/asterisk-box-2-000008f2’
DEBUG[18852] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
4- Debug on Asterisk Box-3 :
DEBUG[31241] chan_sip.c: Header 0 [ 35]: SIP/2.0 480 Temporarily Unavailable
DEBUG[31241] chan_sip.c: Header 7 [101]: Reason: SIP;cause=408;text=“Request Timeout”;icodetext=“NoUserResponding”;iintcode=12013;isubsystem=9
DEBUG[14998] channel.c: Hanging up channel ‘SIP/to-telcooperator-00042473’
DEBUG[14998] app_dial.c: Exiting with DIALSTATUS=CONGESTION.
My questions :
- Can some have an explication why it’s FAILED, is it a problem with our Telco provider or configuration problem with my asterisk
- how to debug PRI with a dedicated channel, or a tool to learn a file debug.
Best Rgds,