Call failed with predictif application

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 + 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 :
  • 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:
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
SDP Session Name: Asterisk PBX
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 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
Externrefresh: 10
Internal IP: x.x.x.x:5060
STUN server:

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 :

  • asterisk/dahdi-channels.conf :
    channel =>1-15,17-31

  • My config iax2 on Asterisk Box-2 :
    iaxthreadcount = 10
    iaxmaxthreadcount = 100

type = peer
username = predictif-users
secret = YTRESDF
host = y.y.y.y
qualify = yes
trunk = yes
requirecalltoken = no
disallow = all
allow = g729
jitterbuffer = yes

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,

The telco seems to be telling you that the called user didn’t answer. At first sight that is a problem with the person being called. If it is a technical problem, it is downstream of Asterisk.